| 2622ea73e33bf4269dcccff89a7ba224a80975b9 |
24-Feb-2017 |
Chih-Hung Hsieh <chh@google.com> |
Leave only an empty top level OWNERS file.
We should not copy OWNERS files from upstream,
or the owners should be registered in Gerrit Code Review.
Bug: 33166666
Test: default build targets
Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
WNERS
pp/webrtc/OWNERS
pp/webrtc/androidtests/OWNERS
pp/webrtc/java/android/org/webrtc/OWNERS
pp/webrtc/java/jni/OWNERS
pp/webrtc/objc/OWNERS
pp/webrtc/objctests/OWNERS
uild/OWNERS
edia/webrtc/OWNERS
|
| fcfc804e436502d49b2176fec1f40dce3585527f |
14-Jan-2016 |
kjellander <kjellander@webrtc.org> |
Eliminate defines in talk/
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions.
Remove no longer used defines from talk/build/common.gypi due to
previously migrated sources (into webrtc/p2p and webrtc/libjingle).
When this is rolled into Chromium, we can also clean up the platform
defines in
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp
NOTRY=True
BUG=webrtc:5420
TESTED=Ran all compile trybots with --clobber flag.
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1588453005
Cr-Commit-Position: refs/heads/master@{#11254}
uild/common.gypi
edia/base/executablehelpers.h
edia/base/mediaengine.h
edia/base/videocapturer.cc
edia/devices/devicemanager.cc
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/devices/v4llookup.h
edia/devices/videorendererfactory.h
edia/webrtc/webrtcvoiceengine.cc
|
| 3542013f587f0858fb24fa8e554ec3c01a323da8 |
14-Jan-2016 |
sprang <sprang@webrtc.org> |
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
Reason for revert:
We're getting boringssl version conflicts. Reverting for now.
Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org, henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}
TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381
Review URL: https://codereview.webrtc.org/1586183002
Cr-Commit-Position: refs/heads/master@{#11253}
pp/webrtc/peerconnection_unittest.cc
|
| 31c8d2eac5aec977f584ab0ae5a1d457d674f101 |
14-Jan-2016 |
Torbjorn Granlund <torbjorng@google.com> |
Update with new default boringssl no-aes cipher suites. Re-enable tests.
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
BUG=webrtc:5381
R=davidben@webrtc.org, henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1550773002 .
Cr-Commit-Position: refs/heads/master@{#11250}
pp/webrtc/peerconnection_unittest.cc
|
| 688e308a353c27803a66803230235638f4dd1f2b |
14-Jan-2016 |
aluebs <aluebs@webrtc.org> |
Re-land: "Use an explicit identifier in Config"
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Original CL: https://codereview.webrtc.org/1538643004/
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1589573004
Cr-Commit-Position: refs/heads/master@{#11248}
ibjingle.gyp
|
| 268493a96b93d6a11a595b3272c5a4cd7a1fdc47 |
14-Jan-2016 |
nisse <nisse@webrtc.org> |
Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
Reason for revert:
These changes broke chrome.
Need to temporarily keep methods InitToEmptyBuffer, InitToBlack, CreateEmptyFrame with old but ignored arguments for pixel_width and pixel_height. Then update chrome, and delete the old methods in a separate cl.
Original issue's description:
> Delete remnants of non-square pixel support from cricket::VideoFrame.
>
> If ever needed, add some aspect ratio parameter, without pixel_width
> and pixel_height arguments cluttering commonly used functions.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/709513d4133107d5c02aed34a5ee99444c4d4e25
> Cr-Commit-Position: refs/heads/master@{#11243}
TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1583223002
Cr-Commit-Position: refs/heads/master@{#11246}
pp/webrtc/videotrack_unittest.cc
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 709513d4133107d5c02aed34a5ee99444c4d4e25 |
14-Jan-2016 |
nisse <nisse@webrtc.org> |
Delete remnants of non-square pixel support from cricket::VideoFrame.
If ever needed, add some aspect ratio parameter, without pixel_width
and pixel_height arguments cluttering commonly used functions.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1586613002
Cr-Commit-Position: refs/heads/master@{#11243}
pp/webrtc/videotrack_unittest.cc
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 2d110be77f14cab0bb51efe8b61d9c7a967d04cb |
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.
Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}
TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1588693002
Cr-Commit-Position: refs/heads/master@{#11241}
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| fca54f41ad1b5b2189d123fe8e97f3ff9b457336 |
13-Jan-2016 |
tommi <tommi@webrtc.org> |
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
Reason for revert:
Reverting due to problem with roll:
/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
-> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
configs -= [ "//build/config/clang:find_bad_constructs" ]
^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@
Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}
TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1586563003
Cr-Commit-Position: refs/heads/master@{#11239}
ibjingle.gyp
|
| 306efadffab8e9dbd494f02d75c78c46ee95689f |
13-Jan-2016 |
kjellander <kjellander@webrtc.org> |
Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
BUG=webrtc:4963
TBR=pbos@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1577233005
Cr-Commit-Position: refs/heads/master@{#11237}
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 25249d92d3cf105bcc7b684c8924ccdbc9afcb93 |
13-Jan-2016 |
aluebs <aluebs@webrtc.org> |
Use an explicit identifier in Config
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Review URL: https://codereview.webrtc.org/1538643004
Cr-Commit-Position: refs/heads/master@{#11231}
ibjingle.gyp
|
| e591f9377f33f3f725a30faecd1bef1a71fa6b99 |
13-Jan-2016 |
deadbeef <deadbeef@webrtc.org> |
Storing raw audio sink for default audio track.
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1551813002
Cr-Commit-Position: refs/heads/master@{#11230}
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 6955870806624479723addfae6dcf5d13968796c |
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
pp/webrtc/mediastreaminterface.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
pp/webrtc/webrtcsdp.cc
edia/base/audiorenderer.h
edia/base/codec.cc
edia/base/codec.h
edia/base/fakemediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| 3e1cfa7edba8081faada275683b3d1fc71f37ac7 |
12-Jan-2016 |
nisse <nisse@webrtc.org> |
Delete unused method webrtc::VideoRendererInterface::SetSize.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1582493002
Cr-Commit-Position: refs/heads/master@{#11223}
pp/webrtc/mediastreaminterface.h
|
| 127782bbb11da802ad2494939965cc50119ecd38 |
12-Jan-2016 |
nisse <nisse@webrtc.org> |
Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1581583002
Cr-Commit-Position: refs/heads/master@{#11218}
pp/webrtc/videosource.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
edia/base/videorenderer.h
|
| b2328d11dcc86fba1661ee3fa0d51fc126939764 |
12-Jan-2016 |
aluebs <aluebs@webrtc.org> |
Remove additional channel constraints when Beamforming is enabled in AudioProcessing
The general constraints on number of channels for AudioProcessing is:
num_in_channels == num_out_channels || num_out_channels == 1
When Beamforming is enabled and additional constraint was added forcing:
num_out_channels == 1
This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo.
Review URL: https://codereview.webrtc.org/1571013002
Cr-Commit-Position: refs/heads/master@{#11215}
edia/webrtc/fakewebrtcvoiceengine.h
|
| a7446d2a50167602b04f58c917f5075ad5e494dc |
12-Jan-2016 |
Guo-wei Shieh <guoweis@webrtc.org> |
Change DTLS default from 1.0 to 1.2 for webrtc.
This changes for standalone webrtc applications.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1548733002 .
Cr-Commit-Position: refs/heads/master@{#11211}
pp/webrtc/peerconnectioninterface.h
|
| 27ed3cc28cf950456a0c66d7a10656a96832fedd |
11-Jan-2016 |
lally <lally@chromium.org> |
SCTP: Stopped accepting SSRCs higher than max.
Seems to fix asan-related crash.
BUG=https://code.google.com/p/chromium/issues/detail?id=570261
Review URL: https://codereview.webrtc.org/1571853002
Cr-Commit-Position: refs/heads/master@{#11205}
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
|
| f475d365a25036725c3f545f57de59d2cc902d17 |
09-Jan-2016 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Properly handle different transports having different SSL roles.
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
|
| 25702cb1628941427fa55e528f53483f239ae011 |
08-Jan-2016 |
pkasting <pkasting@chromium.org> |
Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks
All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1534193008
Cr-Commit-Position: refs/heads/master@{#11191}
edia/base/codec.cc
edia/base/codec.h
|
| 37ebcf0ce5ad1685bcf659ea75960beb96019647 |
08-Jan-2016 |
phoglund <phoglund@webrtc.org> |
Reland "Add APK targets to build libjingle tests for Android."
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/
This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391.
We have made more preparations downstream, so this should work now. Original CL by perkj@.
BUG=webrtc:2365
The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/
Review URL: https://codereview.webrtc.org/1570513004
Cr-Commit-Position: refs/heads/master@{#11186}
pp/webrtc/java/jni/jni_onload.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/DEPS
pp/webrtc/test/androidtestinitializer.cc
pp/webrtc/test/androidtestinitializer.h
pp/webrtc/webrtcsdp_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
|
| fbeb97e01f02a528cce02f076942a779195270a5 |
08-Jan-2016 |
perkj <perkj@webrtc.org> |
Fix clang warning in peerconnection_jni.cc
TEST= export GYP_DEFINES="OS=android clang=1" ...
ninja -C out/Debug AppRTCDemo
BUG=webrtc:5399
Review URL: https://codereview.webrtc.org/1561073005
Cr-Commit-Position: refs/heads/master@{#11181}
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
|
| 893505d0fb41a840be5e4a44a1250dba83d79bf5 |
08-Jan-2016 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Adding unit test to ensure TURN server priorities are unique.
BUG=webrtc:5209
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1570563002 .
Cr-Commit-Position: refs/heads/master@{#11177}
pp/webrtc/peerconnection_unittest.cc
|
| e5ba13bc09a17ada23e0ed6e6f0eb7f3476a1ed0 |
08-Jan-2016 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Adding a way for a Java RtpSender to set a track without taking ownership.
This means that the track will still have a reference count after the
PeerConnection and RtpSender have been destroyed.
R=glaznev@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1566103003 .
Cr-Commit-Position: refs/heads/master@{#11176}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/RtpSender.java
|
| 13f61dfea59a546e4e0081eb79e38c542ec51cf6 |
04-Jan-2016 |
Peter Boström <pbos@webrtc.org> |
Move fake-handle frame creation into test target.
Renames CreateFakeNativeHandleFrame to FakeNativeHandle::CreateFrame and
moves into test.gyp target 'fake_video_frames' which contains previous
frame_generator target.
Removes unused warnings from includers of
webrtc/test/fake_texture_frame.h which did not use the function above.
BUG=webrtc:5398
R=kjellander@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1554223002 .
Cr-Commit-Position: refs/heads/master@{#11149}
ibjingle_tests.gyp
|
| 60ca31bf5d206ff01b5441639806f7303365e162 |
04-Jan-2016 |
kjellander <kjellander@webrtc.org> |
Roll chromium_revision d66326c..4df108a (367167:367307)
The changes in https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a/build/common.gypi
enables a lot more warnings, which have been disabled/fixed in this CL.
See tracking bugs for remaining work.
Change log: https://chromium.googlesource.com/chromium/src/+log/d66326c..4df108a
Full diff: https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a
Changed dependencies:
* src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/fee7f1e..6d0c448
* src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/b8dd754..8a7662a
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a/DEPS
No update to Clang.
BUG=webrtc:5397, webrtc:5398, webrtc:5399
TBR=hta@webrtc.org, perkj@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1553033002
Cr-Commit-Position: refs/heads/master@{#11147}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/objc/avfoundationvideocapturer.h
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/devices/carbonvideorenderer.cc
edia/devices/carbonvideorenderer.h
|
| 0c7e9f540b282d60b94081f601a1694054d8646e |
29-Dec-2015 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Removing webrtc::PortAllocatorFactoryInterface.
ICE servers are now passed directly into PortAllocator,
making PortAllocatorFactoryInterface redundant. This CL also
moves SetNetworkIgnoreMask to PortAllocator.
R=phoglund@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1520963002 .
Cr-Commit-Position: refs/heads/master@{#11139}
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/portallocatorfactory.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
ibjingle.gyp
|
| 3f7219be700df3fea85193e8d541e7f90a1c3ce6 |
29-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing issue where description contains empty ICE ufrag/pwd.
The issue occurred when deserializing and then serializing a rejected
content description, which doesn't have the ICE ufrag/pwd in the first
place.
BUG=webrtc:5105
Review URL: https://codereview.webrtc.org/1534363002
Cr-Commit-Position: refs/heads/master@{#11134}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| e6bf587259da23e96a8de0957b172fd74c36c3c6 |
21-Dec-2015 |
nisse <nisse@webrtc.org> |
Deleted VideoCapturer::screencast_max_pixels, together with
VideoChannel::GetScreencastMaxPixels and VideoChannel::GetScreencastFps.
Unused in webrtc, also unused in everything indexed by google and chromium code search. With the exception of the magicflute plugin, which I'm told doesn't matter.
Review URL: https://codereview.webrtc.org/1532133002
Cr-Commit-Position: refs/heads/master@{#11108}
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 2f042f26a3d0c062c43dc553058a286bd4dd8f19 |
20-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Roll chromium_revision 1b6c421..db567a8 (365999:366304)
I had to disable some Dtls12Both tests failing under MSan (see bug).
Notice those errors started happening in the range of
https://boringssl.googlesource.com/boringssl.git/+log/afd565f..9f897b2
while this CL brings in an even newer BoringSSL (that still has the same problem).
Change log: https://chromium.googlesource.com/chromium/src/+log/1b6c421..db567a8
Full diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8
Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/afd565f..afe57cb
* src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/1019e45..1ccbf8f
* src/third_party/nss: https://chromium.googlesource.com/chromium/deps/nss.git/+log/a676aa0..aee1b12
DEPS diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8/DEPS
No update to Clang.
NOTRY=True
BUG=webrtc:5381
TBR=torbjorng@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1533253002
Cr-Commit-Position: refs/heads/master@{#11095}
pp/webrtc/peerconnection_unittest.cc
|
| a4df27b6713583045e51e20c4eb93718d15ca33e |
19-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
Reason for revert:
Compile error on Android needs to be fixed before relanding.
Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}
TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1537213002
Cr-Commit-Position: refs/heads/master@{#11094}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| f4f5cb09277d5ef6aeac8341e5f54a055867803a |
19-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/
The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
NOTRY=true
TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1541633002
Cr-Commit-Position: refs/heads/master@{#11093}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| bd7d8f7e2b824a887aa12236cb6185d446d7da61 |
19-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding a MediaStream parameter to createSender.
This will allow an app to create senders with the same stream id,
without SDP munging.
Review URL: https://codereview.webrtc.org/1538673002
Cr-Commit-Position: refs/heads/master@{#11092}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
|
| 36d4c545007129446e551c45c17b25377dce89a4 |
18-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.
Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}
TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1533913004
Cr-Commit-Position: refs/heads/master@{#11087}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| b7d9a97ce41022e984348efb5f28bf6dd6c6b779 |
18-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.
BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1406903002 .
Cr-Commit-Position: refs/heads/master@{#11084}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| ae2c5ad12afc8cc29fe9c59dea432b697b871a87 |
18-Dec-2015 |
ivoc <ivoc@webrtc.org> |
Added option to specify a maximum file size when recording an AEC dump.
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.
BUG=webrtc:4741
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1413483003
Cr-Commit-Position: refs/heads/master@{#11081}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 88518a22c62ccb7989a0e10d43bea1a63cdfcd09 |
18-Dec-2015 |
perkj <perkj@webrtc.org> |
Use NV21 instead of YUV12 and clean up.
BUG=webrtc:5375
Review URL: https://codereview.webrtc.org/1530843002
Cr-Commit-Position: refs/heads/master@{#11079}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
|
| 48477c1c6a6e4f70dd3a4a559b5235108f8709ed |
18-Dec-2015 |
perkj <perkj@webrtc.org> |
MediaCodecVideoEncoder, set timestamp on the encoder surface when drawing a texture.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1523843006
Cr-Commit-Position: refs/heads/master@{#11078}
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/java/android/org/webrtc/EglBase14.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 77fa59d78923815ef7403bd738784e9c0be24c54 |
18-Dec-2015 |
guoweis <guoweis@webrtc.org> |
Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1537683003
Cr-Commit-Position: refs/heads/master@{#11076}
ession/media/srtpfilter.cc
|
| 4638331fd8857b263bb65f12dbf5e1f7005e1a9a |
18-Dec-2015 |
guoweis <guoweis@webrtc.org> |
DTLS-SRTP set up is bypassed when the channel has been writable.
This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied.
We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL.
BUG=568734
Review URL: https://codereview.webrtc.org/1532543003
Cr-Commit-Position: refs/heads/master@{#11075}
pp/webrtc/peerconnection_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/srtpfilter.cc
|
| 0eb15ed7b806125774bd13fb214aeb403e2c6857 |
17-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/test/fakedtlsidentitystore.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/channel.cc
ession/media/channel_unittest.cc
ession/media/mediasession_unittest.cc
|
| a54a0801121e05f797e514731cc5c9bad2f5e597 |
17-Dec-2015 |
honghaiz <honghaiz@webrtc.org> |
Add ufrag to the ICE candidate signaling.
On the receiving side, if a candidate arrives with an old ufrag, it will be dropped. If it contains a new frag that has never seen before, it will hold the ufrag and create connections, although those connections are not pingable until the ICE credentials are received.
This could avoid a bunch of ICE generation issues.
BUG=webrtc:5138,webrt:5292
Review URL: https://codereview.webrtc.org/1498993002
Cr-Commit-Position: refs/heads/master@{#11060}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 7cae30cbe1200854bbe26205ab53d0f418c8d443 |
16-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Disable warnings failing when using Clang on Windows.
This makes it possible to build WebRTC using Clang on Windows.
Depends on https://codereview.webrtc.org/1524703006/
BUG=webrtc:5360, webrtc:5366
NOTRY=True
Review URL: https://codereview.webrtc.org/1522223002
Cr-Commit-Position: refs/heads/master@{#11058}
ibjingle_tests.gyp
|
| 672aba3f57061e33dd802d9a391c54bdfed952c3 |
16-Dec-2015 |
perkj <perkj@webrtc.org> |
Fix error prone code in VideoCapturerAndroid
BUG=webrtc:5282
Review URL: https://codereview.webrtc.org/1486423003
Cr-Commit-Position: refs/heads/master@{#11046}
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 66085beef83c790a69666b9be8a74bb2eee44fab |
16-Dec-2015 |
peah <peah@webrtc.org> |
Bugfix that fixes the error where the audio processing module is called
using the wrong sample rate for the render signal.
The CL is basically a partial revert of the related changes done on
output_mixer.cc in the CL https://codereview.webrtc.org/1234463003.
The CL also reverts the removal of the input_sample_rate_hz() method
that was removed as part of the CL
https://codereview.webrtc.org/1379123002 (as it was at that point
no longer used).
It should be noted that this CL turns off the effect of the
IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are
used. While it may be possible to solve that by adding upsampling after
the API call, that approach was discarded due to that:
-That would add extra processing in the echo path, leading to possible
AEC performance reduction.
-That would add extra complexity for the mobile case.
-That would only patch the intelligibility enhancer operation as the
proper way to do such an operation is within APM.
-The intelligibility enhancer is not active by default anywhere.
BUG=webrtc:5237
Review URL: https://codereview.webrtc.org/1525173002
Cr-Commit-Position: refs/heads/master@{#11045}
edia/webrtc/fakewebrtcvoiceengine.h
|
| eb45981165f982dd51425fad5ecb7ea9619063d3 |
16-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Restoring behavior where PeerConnection tracks changes to MediaStreams.
If a MediaStream is added to a PeerConnection, and later a track
is added to the MediaStream, a new RtpSender will now be created for
that track, and it will appear in subsequent offers.
Similarly, removed tracks will remove RtpSenders.
BUG=webrtc:5265
Review URL: https://codereview.webrtc.org/1507973003
Cr-Commit-Position: refs/heads/master@{#11040}
pp/webrtc/mediastreamobserver.cc
pp/webrtc/mediastreamobserver.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
ibjingle.gyp
|
| 44f0819978c2ba1f765835bca91e3243eb9f638b |
16-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing bug where "mid" wasn't preserved across re-offers.
Review URL: https://codereview.webrtc.org/1529673002
Cr-Commit-Position: refs/heads/master@{#11039}
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 51254331ccb3838b03ed0c630f7e3d5d402d1919 |
15-Dec-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Refactor renderers to allow apps to inject custom shaders
This CL:
* Abstracts the functions in GlRectDrawer to an interface.
* Adds viewport location as argument to the draw() functions, because this information may be needed by some shaders. This also moves the responsibility of calling GLES20.glViewport() to the drawer.
* Moves uploadYuvData() into a separate helper class.
* Adds new SurfaceViewRenderer.init() function and new VideoRendererGui.create() function that takes a custom drawer as argument. Each YuvImageRenderer in VideoRendererGui now has their own drawer instead of a common one.
BUG=b/25694445
R=nisse@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1520243003 .
Cr-Commit-Position: refs/heads/master@{#11031}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/GlRectDrawer.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 32d989b3f2d168327ed43d0e4c493550ccee4179 |
15-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Disable transport sequence numbers for audio.
Since this isn't fully wired up yet it shouldn't be part of the
SendSideBwe experiment yet.
BUG=webrtc:5263
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1523283002 .
Cr-Commit-Position: refs/heads/master@{#11029}
edia/webrtc/webrtcvoiceengine.cc
|
| 6eca7e3c371383020095ba346e1ac70f38a8c0fd |
15-Dec-2015 |
tommi <tommi@webrtc.org> |
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
Additionally:
* Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack.
* AddSink/RemoveSink are now on all audio sources (like they are for video sources).
While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state.
BUG=chromium:569526
Review URL: https://codereview.webrtc.org/1522903002
Cr-Commit-Position: refs/heads/master@{#11026}
pp/webrtc/audiotrack.cc
pp/webrtc/audiotrack.h
pp/webrtc/localaudiosource.h
pp/webrtc/mediastream_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/remoteaudiotrack.cc
pp/webrtc/remoteaudiotrack.h
pp/webrtc/rtpreceiver.cc
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/videosource.h
pp/webrtc/videosource_unittest.cc
pp/webrtc/videosourceproxy.h
pp/webrtc/videotrack_unittest.cc
ibjingle.gyp
|
| 9638143033f27a3a58d68eb0183eec71350c5479 |
15-Dec-2015 |
perkj <perkj@webrtc.org> |
Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ )
Reason for revert:
Clients have been updated.
Original issue's description:
> Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ )
>
> Reason for revert:
> Revert due breaking other clients.
>
> Original issue's description:
> > Made EglBase an abstract class and cleaned up.
> > Adds EglBase10 that implemenents EglBase for EGL 1.0
> >
> > BUG=webrtc:4993
> > TBR=glaznew@webrtc.org
> >
> > Committed: https://crrev.com/3207916f35ded33f586774e2c98d4d0089fe3c6e
> > Cr-Commit-Position: refs/heads/master@{#11011}
>
> TBR=magjed@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4993
>
> Committed: https://crrev.com/e22e1cb399748112f308b488e7535754ef6b807d
> Cr-Commit-Position: refs/heads/master@{#11013}
TBR=magjed@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1522303004
Cr-Commit-Position: refs/heads/master@{#11024}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/EglBase10.java
pp/webrtc/java/android/org/webrtc/EglBase14.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
|
| 158879305bf5910c0b9e3630a073324a048b59ef |
15-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing flaky LocalP2PTestSctpDataChannel test.
SCTP data channels are closed asynchronously in-band, unlike RTP
data channels, so the test must be slightly modified.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1527833003
Cr-Commit-Position: refs/heads/master@{#11017}
pp/webrtc/peerconnection_unittest.cc
|
| c9be00797edf9a12ff88c81bb56194c74dcacf7f |
15-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing and re-enabling some flaky PeerConnection tests.
BUG=webrtc:3362
Review URL: https://codereview.webrtc.org/1512763003
Cr-Commit-Position: refs/heads/master@{#11016}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| bd292465ee6d8219b04f17e2ffc0790313167f01 |
15-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Free SCTP data channels asynchronously in PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1513143003/ )
Original issue's description:
> Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ )
>
> Reason for revert:
> Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
>
> Original issue's description:
> > Free SCTP data channels asynchronously in PeerConnection.
> >
> > This is needed so that the data channel isn't deleted while one of its
> > own methods is on the call stack.
> >
> > BUG=565048
> >
> > Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> > Cr-Commit-Position: refs/heads/master@{#10923}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=565048
>
> Committed: https://crrev.com/a1f567ae9012a8de573b5bde492dd9ca0d17f137
> Cr-Commit-Position: refs/heads/master@{#10977}
BUG=565048
Review URL: https://codereview.webrtc.org/1516943002
Cr-Commit-Position: refs/heads/master@{#11015}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| e22e1cb399748112f308b488e7535754ef6b807d |
14-Dec-2015 |
perkj <perkj@webrtc.org> |
Revert of Made EglBase an abstract class and cleaned up. (patchset #4 id:60001 of https://codereview.webrtc.org/1526463002/ )
Reason for revert:
Revert due breaking other clients.
Original issue's description:
> Made EglBase an abstract class and cleaned up.
> Adds EglBase10 that implemenents EglBase for EGL 1.0
>
> BUG=webrtc:4993
> TBR=glaznew@webrtc.org
>
> Committed: https://crrev.com/3207916f35ded33f586774e2c98d4d0089fe3c6e
> Cr-Commit-Position: refs/heads/master@{#11011}
TBR=magjed@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1522073002
Cr-Commit-Position: refs/heads/master@{#11013}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/EglBase10.java
pp/webrtc/java/android/org/webrtc/EglBase14.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
|
| 3207916f35ded33f586774e2c98d4d0089fe3c6e |
14-Dec-2015 |
perkj <perkj@webrtc.org> |
Made EglBase an abstract class and cleaned up.
Adds EglBase10 that implemenents EglBase for EGL 1.0
BUG=webrtc:4993
TBR=glaznew@webrtc.org
Review URL: https://codereview.webrtc.org/1526463002
Cr-Commit-Position: refs/heads/master@{#11011}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/EglBase10.java
pp/webrtc/java/android/org/webrtc/EglBase14.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
|
| bc14164aad254e72ce4d1e381b912b7d3acf5391 |
14-Dec-2015 |
stefan <stefan@webrtc.org> |
Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
Reason for revert:
Breaks bots.
Original issue's description:
> Add APK targets to build libjingle_peerconnection_unittests for Android.
>
> BUG=webrtc:2365
>
> The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/
>
> Committed: https://crrev.com/a78c0211fd50369a75a962385db6163bd8ded239
> Cr-Commit-Position: refs/heads/master@{#11007}
TBR=kjellander@webrtc.org,tommi@webrtc.org,perkj@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2365
Review URL: https://codereview.webrtc.org/1521993002
Cr-Commit-Position: refs/heads/master@{#11009}
pp/webrtc/java/jni/jni_onload.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/DEPS
pp/webrtc/test/androidtestinitializer.cc
pp/webrtc/test/androidtestinitializer.h
pp/webrtc/webrtcsdp_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
|
| a78c0211fd50369a75a962385db6163bd8ded239 |
14-Dec-2015 |
perkj <perkj@webrtc.org> |
Add APK targets to build libjingle_peerconnection_unittests for Android.
BUG=webrtc:2365
The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/
Review URL: https://codereview.webrtc.org/1511633002
Cr-Commit-Position: refs/heads/master@{#11007}
pp/webrtc/java/jni/jni_onload.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/DEPS
pp/webrtc/test/androidtestinitializer.cc
pp/webrtc/test/androidtestinitializer.h
pp/webrtc/webrtcsdp_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
|
| 17821db19702aca15d0d93cb60515ca70823fad7 |
14-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Wire up bandwidth limitation info to GetStats and adapt_reason.
The input resolution (output from video_adapter) can be further scaled down or higher video layer(s) can be dropped due to bitrate constraints.
BUG=webrtc:4112
Review URL: https://codereview.webrtc.org/1502173002
Cr-Commit-Position: refs/heads/master@{#11006}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 1d5c19d23eb0d0007efd1e80f60f5409c4b25e25 |
14-Dec-2015 |
tommi <tommi@webrtc.org> |
Address comments from code review 1505253004
(https://codereview.webrtc.org/1505253004/)
BUG=
Review URL: https://codereview.webrtc.org/1523603002
Cr-Commit-Position: refs/heads/master@{#11002}
edia/base/mediachannel.h
|
| 4759bfb2a4fa55f440d947f8f71ea033d85a2215 |
14-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Roll chromium_revision 7de03ed..4bc4277 (364770:364953)
Change log: https://chromium.googlesource.com/chromium/src/+log/7de03ed..4bc4277
Full diff: https://chromium.googlesource.com/chromium/src/+/7de03ed..4bc4277
Changed dependencies:
* src/third_party/usrsctp/usrsctplib: Moved from
https://chromium.googlesource.com/external/usrsctplib.git/+/36444a9
to https://chromium.googlesource.com/external/github.com/sctplab/usrsctp/+/c60ec8b
DEPS diff: https://chromium.googlesource.com/chromium/src/+/7de03ed..4bc4277/DEPS
No update to Clang.
TBR=
Review URL: https://codereview.webrtc.org/1521303003
Cr-Commit-Position: refs/heads/master@{#11001}
ibjingle.gyp
|
| cb95f54ee469cd9682ffecd404d37c5e4d58edb0 |
12-Dec-2015 |
Tommi <tommi@webrtc.org> |
Remove pointless move() to fix build on clang/win.
Fixes:
..\..\third_party\libjingle\source\talk\app\webrtc\remoteaudiosource.cc(100,15)
: error: moving a temporary object prevents copy elision
[-Werror,-Wpessimizing-move]
ssrc, std::move(rtc::scoped_ptr<AudioSinkInterface>(new Sink(this))));
^
..\..\third_party\libjingle\source\talk\app\webrtc\remoteaudiosource.cc(100,15)
: note: remove std::move call here
ssrc, std::move(rtc::scoped_ptr<AudioSinkInterface>(new Sink(this))));
^~~~~~~~~~
R=thakis@chromium.org
TBR=thakis@chromium.org
Review URL: https://codereview.webrtc.org/1517253004 .
Cr-Commit-Position: refs/heads/master@{#10999}
pp/webrtc/remoteaudiosource.cc
|
| f888bb58da04c5095759b5ec7ce2e1fa2cd414fd |
12-Dec-2015 |
Tommi <tommi@webrtc.org> |
Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1505253004 .
Cr-Commit-Position: refs/heads/master@{#10995}
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnection.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/remoteaudiotrack.cc
pp/webrtc/remoteaudiotrack.h
pp/webrtc/rtpreceiver.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| 04e9146e58bd68339b15ad651c9ee593d781e040 |
11-Dec-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Discard old-generation candidates when ICE restarts
The existing code only do so on the controlled side.
BUG=webrtc:5291
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1496693002 .
Cr-Commit-Position: refs/heads/master@{#10993}
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| 822bdf978435b8eba9343ea96e9a9bc54b9c7df0 |
11-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Remove cricket::VideoEncoderConfig.
BUG=webrtc:5332
R=noahric@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1512853007 .
Cr-Commit-Position: refs/heads/master@{#10991}
pp/webrtc/webrtcsession.cc
edia/base/codec.h
edia/base/codec_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 71f5a9a37750d6ccea110028e3154ee90334ba6d |
11-Dec-2015 |
Per <perkj@chromium.org> |
This cl change VideoCaptureAndroid to handle CVO the same way when capturing to texture as when using ordinary byte buffers.
Ie, rotation is applied in C++ in the VideoFrameFactory is apply_rotation_ is set. If not, rotation is sent in RTP.
BUG=webrtc:4993
R=nisse@chromium.org
Review URL: https://codereview.webrtc.org/1493913007 .
Cr-Commit-Position: refs/heads/master@{#10986}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
|
| cf846ad60adcfe11740d58a097fbdc8e02b2839b |
11-Dec-2015 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Adding stub files needed for https://codereview.webrtc.org/1507973003/
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1519683002 .
Cr-Commit-Position: refs/heads/master@{#10981}
pp/webrtc/mediastreamobserver.cc
pp/webrtc/mediastreamobserver.h
|
| 7c73bdbd82956729ee2274318a451a481164f0c6 |
11-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor.
Updating blacklists as well.
Review URL: https://codereview.webrtc.org/1508683004
Cr-Commit-Position: refs/heads/master@{#10980}
pp/webrtc/peerconnection_unittest.cc
|
| a1f567ae9012a8de573b5bde492dd9ca0d17f137 |
10-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Free SCTP data channels asynchronously in PeerConnection. (patchset #3 id:40001 of https://codereview.webrtc.org/1492383002/ )
Reason for revert:
Breaks WebrtcTransportTest.DataStream, due to different rtc::Thread implementation on Chromium.
Original issue's description:
> Free SCTP data channels asynchronously in PeerConnection.
>
> This is needed so that the data channel isn't deleted while one of its
> own methods is on the call stack.
>
> BUG=565048
>
> Committed: https://crrev.com/386869247f28e72a00307a1b5c92465eea343ad2
> Cr-Commit-Position: refs/heads/master@{#10923}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=565048
Review URL: https://codereview.webrtc.org/1513143003
Cr-Commit-Position: refs/heads/master@{#10977}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| 796cfaf7f76aa740cc7f4bb2c94f88637e475324 |
10-Dec-2015 |
perkj <perkj@webrtc.org> |
Add VideoCodec::PreferDecodeLate
The purpose is so that a decoder (Android) that only have a limited number of output buffers can make sure that decoding is done just before the frame is needed.
Removed unused iSupportsRenderTiming and the settings structs since it was not used.
Added VCMReceiver::FrameForDecoding unit test for the case when PreferDecodeLate is set.
Note that this does not change the current behaviour. We actually currently always decode frames late. This cl is to make sure the behaviour is kept for Android, if the default behaviour is changed.
Review URL: https://codereview.webrtc.org/1428293003
Cr-Commit-Position: refs/heads/master@{#10974}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
|
| c490e01bd1bd4a0d754ed5f746b95ac03136346f |
10-Dec-2015 |
nisse <nisse@webrtc.org> |
Implement NativeToI420Buffer in C++, calling java SurfaceTextureHelper, new method .textureToYUV, to
do the conversion using an opengl fragment shader.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1460703002
Cr-Commit-Position: refs/heads/master@{#10972}
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
|
| 1387149ad1669365ac05278bf779a407bec08a4e |
09-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding reduced size RTCP configuration down to the video stream level.
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
ession/media/channel.cc
ession/media/mediasession.cc
ession/media/mediasession.h
|
| 434aca8d862a46d0c3b71698a264d0c71d898170 |
09-Dec-2015 |
tommi <tommi@webrtc.org> |
Add empty placeholder files for remote audio tracks.
This is needed for Chromium so that we can roll, update libjingle.gyp and then continue.
BUG=chromium:121673
Review URL: https://codereview.webrtc.org/1514573003
Cr-Commit-Position: refs/heads/master@{#10955}
pp/webrtc/remoteaudiotrack.cc
pp/webrtc/remoteaudiotrack.h
ibjingle.gyp
|
| 7623ce4aeb9130c937ba5836490cbb3a35679e79 |
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.
Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}
BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1510183002 .
Cr-Commit-Position: refs/heads/master@{#10948}
edia/base/videocommon.cc
|
| bda7e0b932fc89598da95496efc8650bc0e2c86c |
09-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing issue with default stream upon setting 2nd remote description.
If a description is set that requires making a default stream, and one
already exists, we'll now keep the existing default audio/video tracks,
rather than destroying them and recreating them. Destroying them caused
the blink MediaStream to go to an "ended" state, which is the root cause
of the bug.
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1469833006
Cr-Commit-Position: refs/heads/master@{#10946}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
|
| d02b0fab76847c72bd45e5a40763255283abe212 |
08-Dec-2015 |
haysc <haysc@webrtc.org> |
Add oldest rotation type option to RTCFileLogger
BUG=
Review URL: https://codereview.webrtc.org/1432753003
Cr-Commit-Position: refs/heads/master@{#10945}
pp/webrtc/objc/RTCFileLogger.mm
pp/webrtc/objc/public/RTCFileLogger.h
|
| 1a9d615cbf93662519748aafc96d1ea23fa1a9e1 |
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Add tracing to public PeerConnection methods.
Adds tracing specifically to Close, for creating streams and also moves
tracing for SetLocal/RemoteDescription from WebRtcSession. Also adding
some tracing in ChannelManager to see what's taking time inside Close.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1509903002 .
Cr-Commit-Position: refs/heads/master@{#10943}
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession.cc
ession/media/channelmanager.cc
|
| 7b2f7627e4241cf0904f63ee6e94eeff3ba9b2e0 |
08-Dec-2015 |
perkj <perkj@webrtc.org> |
Don't call SetPreviewFormat if capturing to textures.
This fix an issue seen on Huawei Y300 where the camera feed is black and white if we capture to textures and setpreviewformat is called.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1502223002
Cr-Commit-Position: refs/heads/master@{#10941}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| edd8fefa9b31f903eefe1e9fcabb09a5d6fc1ad1 |
08-Dec-2015 |
haysc <haysc@webrtc.org> |
Add new view that renders local video using AVCaptureLayerPreview.
BUG=
Review URL: https://codereview.webrtc.org/1497393002
Cr-Commit-Position: refs/heads/master@{#10940}
pp/webrtc/objc/avfoundationvideocapturer.mm
ibjingle.gyp
|
| 246b8171a6fbb4e37a5491679bc595238f81e490 |
08-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Refactor handling of AudioOptions.
- Remove MediaEngineInterface::GetAudioOptions(), SetAudioOptions() and SetSoundDevices().
- Remove the WebRtcVoiceEngine infrastructure for those calls.
BUG=webrtc:4690
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1500633002
Cr-Commit-Position: refs/heads/master@{#10938}
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 8237abf563bf4782ee104408b53cc8e55ce44518 |
08-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}
TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695
Review URL: https://codereview.webrtc.org/1507903005
Cr-Commit-Position: refs/heads/master@{#10937}
edia/base/videocommon.cc
|
| 9f45a45a628100d973111b3aac66dede57454b6a |
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Add tracing to upper-level WebRTC calls.
Adds tracing to WebRtcSession and corresponding BaseChannel calls to see
where time is spent better.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1505023003 .
Cr-Commit-Position: refs/heads/master@{#10934}
pp/webrtc/webrtcsession.cc
edia/webrtc/webrtcvideoengine2.cc
ession/media/channel.cc
|
| 03ef053202bc5d5ab43460eebf5403232f157646 |
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1506773002 .
Cr-Commit-Position: refs/heads/master@{#10926}
edia/base/videocommon.cc
|
| 386869247f28e72a00307a1b5c92465eea343ad2 |
08-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Free SCTP data channels asynchronously in PeerConnection.
This is needed so that the data channel isn't deleted while one of its
own methods is on the call stack.
BUG=565048
Review URL: https://codereview.webrtc.org/1492383002
Cr-Commit-Position: refs/heads/master@{#10923}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| 46ad5426b025eddac8e9232014d347e73d27180e |
07-Dec-2015 |
pbos <pbos@webrtc.org> |
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.
.../webrtc/base/atomicops.h:71:8: note: no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'
Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1505053002
Cr-Commit-Position: refs/heads/master@{#10922}
ession/media/srtpfilter.cc
|
| 6f28cf0b951a9d41246f022f48a6cd035fad151d |
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Implement standalone event tracing in AppRTCDemo.
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1457383002 .
Cr-Commit-Position: refs/heads/master@{#10921}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
ession/media/channel.cc
|
| 84f0970d100e67a1dc4fe9a1b16b7d293302044e |
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Reland of "Create rtc::AtomicInt POD struct."
Relands https://codereview.webrtc.org/1420043008/ with brace initializers
instead of constructors hoping that they won't introduce static
initializers.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1498953002 .
Cr-Commit-Position: refs/heads/master@{#10920}
ession/media/srtpfilter.cc
|
| cd4003f3dfbb95fabdd4cc6a7e4a601bbc06c080 |
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Use @webrtc.org addresses for OWNERS.
Fixes talk/app/webrtc/OWNERS and removes houssainy@google.com from
webrtc/tools/rtcbot/OWNERS.
BUG=
R=andresp@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1505613004 .
Cr-Commit-Position: refs/heads/master@{#10918}
pp/webrtc/OWNERS
|
| cf890bc58eb28d5f1f6ce3f90d4e541983042369 |
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Roll gtest-parallel.
Brings in fixes that save log output to disk instead of piping them
through Python. Should fix problem where output from tests stall for
more than 10 seconds.
Also enabling JsepPeerConnectionP2PTestClient on all platforms again.
BUG=webrtc:5231
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1509463002 .
Cr-Commit-Position: refs/heads/master@{#10917}
pp/webrtc/peerconnection_unittest.cc
|
| 9d69c3f4d99240c27d997c37950b561605d403bd |
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Return a copy of the supported RTP header extensions instead of a reference.
This also renames the method to better reflect what it does.
BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1486123002 .
Cr-Commit-Position: refs/heads/master@{#10910}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
|
| b86d4e4a8dec1eb1b801244a2a97cda66f561d8e |
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
edia/webrtc/webrtcvoiceengine.cc
|
| 03f80ebb8310e5f04ced856f7ec8f14b94a0f47e |
07-Dec-2015 |
nisse <nisse@webrtc.org> |
Refactor EglBase configuration.
Delete EglBase.ConfigType, instead pass arrays of attributes, and define
constant arrays for the common cases.
Both in progress NativeToI420 and extending GlRectDrawer to other shapes (with alpha) needs this.
BUG=b/25694445
Review URL: https://codereview.webrtc.org/1498003002
Cr-Commit-Position: refs/heads/master@{#10908}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/EglBase14.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 1218d7ad2fac035376914bd0649fe99e657b33d3 |
05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Allow remote fingerprint update during a call
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.
TBR=pthatcher@webrtc.org
BUG=webrtc:3618
This is a reland of https://codereview.webrtc.org/1453523002
Review URL: https://codereview.webrtc.org/1505573002 .
Cr-Commit-Position: refs/heads/master@{#10903}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/test/fakedtlsidentitystore.h
ession/media/channel.cc
ession/media/channel.h
ession/media/srtpfilter.h
|
| 86aaa4be8de8f49f91faeefbfd1a23f312898dd2 |
05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Allow remote fingerprint update during a call"
This reverts commit 9c38c2d33fa6d794704d53b18f39d5235439fe63.
This commit somehow is different from what I have in my local copy. Revert and will recommit.
TBR=pthatcher@webrtc.org
BUG=3618
Review URL: https://codereview.webrtc.org/1494373004 .
Cr-Commit-Position: refs/heads/master@{#10902}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/test/fakedtlsidentitystore.h
ession/media/channel.cc
ession/media/channel.h
ession/media/srtpfilter.h
|
| 9c38c2d33fa6d794704d53b18f39d5235439fe63 |
05-Dec-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Allow remote fingerprint update during a call
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.
BUG=webrtc:3618
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1453523002 .
Cr-Commit-Position: refs/heads/master@{#10901}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/test/fakedtlsidentitystore.h
ession/media/channel.cc
ession/media/channel.h
ession/media/srtpfilter.h
|
| 381b4217cb36f434c56e399a852a0a15522a9596 |
04-Dec-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Ping backup connection at a slower rate
and make it configurable from the app.
Changed the decision on whether a connection is pingable:
1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection.
2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate.
Note the default behavior is the same as before.
Also cached the channel state since we are accessing it more often.
BUG=webrtc:5034
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1455033004 .
Cr-Commit-Position: refs/heads/master@{#10900}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/objc/RTCPeerConnectionInterface.mm
pp/webrtc/objc/public/RTCPeerConnectionInterface.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| 9e1b992f74470aecfeb216e26b455982ddc4a6d5 |
04-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Clear old decoders after recreating the receiver.
Prevents UAF when switching decoder capabilities and the
previously-supported decoder is currently being received on.
BUG=chromium:565967
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1490233010 .
Cr-Commit-Position: refs/heads/master@{#10898}
edia/webrtc/webrtcvideoengine2.cc
|
| b572768efbc1e52b97a5ad98932c667956aba4b8 |
04-Dec-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
- Remove calls to VoEDtmf from WVoE/MC.
- Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent().
- Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs().
BUG=webrtc:4690
R=pthatcher@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1491743004 .
Cr-Commit-Position: refs/heads/master@{#10895}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 1a5cf6eab114462fb1111691293e5331ffe23e50 |
04-Dec-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1494693003 .
Cr-Commit-Position: refs/heads/master@{#10889}
edia/base/mediaengine.h
|
| 9cf0c3d4ddfab865dcf924155cc81b763c919a53 |
04-Dec-2015 |
Ivo Creusen <ivoc@webrtc.org> |
Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient.
BUG=webrtc:5231
R=kjellander@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1495853002 .
Cr-Commit-Position: refs/heads/master@{#10887}
pp/webrtc/peerconnection_unittest.cc
|
| 7635684130cc3a071d245b607fddec059002e7fa |
03-Dec-2015 |
tkchin <tkchin@webrtc.org> |
Fix Mac ObjC PeerConnection API compilation.
BUG=webrtc:5287,webrtc:5216
Review URL: https://codereview.webrtc.org/1493003002
Cr-Commit-Position: refs/heads/master@{#10876}
uild/merge_ios_libs.gyp
ibjingle.gyp
ibjingle_tests.gyp
|
| 9462052f32fd777f5437c1e0803246cf6aaa5cdf |
02-Dec-2015 |
honghaiz <honghaiz@webrtc.org> |
In some rare Android systems ConnectivityManager may be null.
Handle this case more gracefully.
BUG=
Review URL: https://codereview.webrtc.org/1490403002
Cr-Commit-Position: refs/heads/master@{#10875}
pp/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java
|
| 3c28d0de95e66905cd0072255b2cf3a1c782bb90 |
02-Dec-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable PeerConnectionEndToEndTest.Call on Mac.
Until the gtest-parallel problem is resolved. This is
needed for CQ stability.
BUG=webrtc:5231
TBR=perkj@webrtc.org,deadbeef@webrtc.org
Review URL: https://codereview.webrtc.org/1499483002 .
Cr-Commit-Position: refs/heads/master@{#10873}
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| 1d63dd0eaa44d13c5ae083200937b18bce2132ae |
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
- Remove the DF_PLAY/DF_SEND flags, only allow sending.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1487393002
Cr-Commit-Position: refs/heads/master@{#10872}
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| ee524f7c02cad35b453d5832ef14640603256e39 |
02-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding Java binding for CreateSender.
Review URL: https://codereview.webrtc.org/1486243002
Cr-Commit-Position: refs/heads/master@{#10871}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
|
| 7e4e01a4413fa98644b94ab9d8a9dccc664f39f2 |
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Add header extension filtering for WebRtcVoiceEngine/MediaChannel.
Rework filtering functionality to be reused for both Audio+Video.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1481963002
Cr-Commit-Position: refs/heads/master@{#10869}
ibjingle_tests.gyp
edia/base/mediachannel.h
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcmediaengine_unittest.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvoiceengine.cc
|
| 2515af28e97213b4a4b89269f6b855378d31e153 |
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1493663002
Cr-Commit-Position: refs/heads/master@{#10868}
edia/webrtc/webrtcvoiceengine.cc
|
| d20d24716697f9ecdee02c279a51018ee95baab2 |
02-Dec-2015 |
perkj <perkj@webrtc.org> |
Fix VideoCaptureAndroid, drop frame when switching camera using textures.
Dropping the first frame intended to fix a problem when switching cameras on N6 when we are capturing to textures but due to a silly bug fixed in this cl the frame was not dropped...
BUG=webrtc:5262
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1489363002
Cr-Commit-Position: refs/heads/master@{#10867}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 226a602ad67b073f10709c25c4f91964985798d7 |
02-Dec-2015 |
perkj <perkj@webrtc.org> |
Fix problem when drawing to the Android Media encoder surface.
Problem seen on N6.
BUG=webrtc:5147
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1491623003
Cr-Commit-Position: refs/heads/master@{#10866}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 40455d6f37fda78ea069a51d95f28994bd736864 |
02-Dec-2015 |
perkj <perkj@webrtc.org> |
This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase.
The reason we want to use EGL14 is to be able to use EGLExt.eglPresentationTimeANDROID when writing textures to MediaEncoder.
BUG=webrtc:4993
TBR=glaznew@webrtc.org
Review URL: https://codereview.webrtc.org/1461083002
Cr-Commit-Position: refs/heads/master@{#10864}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/EglBase14.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
ibjingle.gyp
|
| 41b0798e1171a105404f6bc9dcb591cdc77d659f |
02-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding CreatePeerConnection method that uses new PC Initialize method.
This will let us transition to the new Initialize method in Chromium,
and then get rid of the old one.
Review URL: https://codereview.webrtc.org/1462253002
Cr-Commit-Position: refs/heads/master@{#10860}
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
|
| 0de97f1b748d8238fe3a7ad8d7afb2b6cb456a3e |
01-Dec-2015 |
hbos <hbos@webrtc.org> |
WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest.
Related to issues discussed in the referenced bug but does not solve that bug's main problem.
BUG=webrtc:4776
Review URL: https://codereview.webrtc.org/1485673003
Cr-Commit-Position: refs/heads/master@{#10852}
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer_unittest.cc
|
| cb9792e9f773a40b9f11b79f85b8a495cefb0bef |
01-Dec-2015 |
perkj <perkj@webrtc.org> |
Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M.
Review URL: https://codereview.webrtc.org/1476313002
Cr-Commit-Position: refs/heads/master@{#10850}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
|
| 14f4144a82558ec4da2d4962ef02b23f44967b6a |
01-Dec-2015 |
perkj <perkj@webrtc.org> |
Add helper KeepRefUntilDone.
The callback keeps a reference to an object until the callback goes out of scope.
Review URL: https://codereview.webrtc.org/1487493002
Cr-Commit-Position: refs/heads/master@{#10847}
pp/webrtc/java/jni/native_handle_impl.cc
|
| ee69ed505b6ba4a9dbb47cc927aaf220d661fa06 |
01-Dec-2015 |
glaznev <glaznev@webrtc.org> |
Add separate event for camera freeze.
Review URL: https://codereview.webrtc.org/1479523003
Cr-Commit-Position: refs/heads/master@{#10846}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 70c0e298cb14946a21f698298ae0d6daa73cdfcd |
30-Nov-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable PeerConnectionEndToEndTest.Call for TSan.
Recent flakes:
https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/4565/steps/libjingle_peerconnection_unittest/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/4559/steps/libjingle_peerconnection_unittest/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/4557/steps/libjingle_peerconnection_unittest/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/4549/steps/libjingle_peerconnection_unittest/logs/stdio
BUG=webrtc:4719
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1487823002 .
Cr-Commit-Position: refs/heads/master@{#10845}
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| ae54b835eab068e01c0d2735ad8fcafc9711c91d |
28-Nov-2015 |
magjed <magjed@webrtc.org> |
Android SurfaceViewRenderer: Add resetStatistics() method
Review URL: https://codereview.webrtc.org/1472323003
Cr-Commit-Position: refs/heads/master@{#10833}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 2fe1cb0f0acb66e6f8df47365aac816cb69eb911 |
28-Nov-2015 |
andrew <andrew@webrtc.org> |
Don't overwrite audio stats when they're not available.
Chromium implements AudioProcessorInterface::GetStats(), but other
clients may not. The existing stats were getting overwritten with
default AudioProcessorStats values in that case.
Now, we only overwrite the stats if the track has an
AudioProcessorInterface. Also, move signal level out of
SetAudioProcessingStats() to avoid the "don't set if it's -1" pattern.
Review URL: https://codereview.webrtc.org/1469803004
Cr-Commit-Position: refs/heads/master@{#10831}
pp/webrtc/statscollector.cc
|
| 26c8c91de2db5da06ff337aae48e1d725aa91ab7 |
27-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Using Rent-A-Codec for static Codec access in WVoE/MC.
Mostly moved code around in WebRtcVoiceEngine:
- Added new internal class WebRtcVoiceCodecs for static codec functions and the CodecPrefs.
- ConstructCodecs() -> WebRtcVoiceCodecs::SupportedCodecs().
- FindWebRtcCodec -> WebRtcVoiceCodecs::ToCodecInst().
- WebRtcVoiceMediaChannel::SetRecvCodecsInternal() folded into WebRtcVoiceMediaChannel::SetRecvCodecs() (slight logic change).
- Change to how SetRecPayloadType() is implemented in fakewebrtcvoiceengine.h (lines 460-470).
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1461333002
Cr-Commit-Position: refs/heads/master@{#10819}
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 727dbc2968c8761a3150faebba254155fb042530 |
26-Nov-2015 |
Per <perkj@chromium.org> |
VideoCapturerAndroid - allow lower frame rate in bad lightning
Insted of using a fixed frame rate, we allow the camera to use a lower frame rate. The camera will choose depending on lightning condition.
TESTED= In a room with low light on N5, N6 N7, Galaxy 4.
BUG=webrtc:5262
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1479563004 .
Cr-Commit-Position: refs/heads/master@{#10807}
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
|
| 598242a583564c5816a4b3b3c93f5cccf2395a17 |
26-Nov-2015 |
Per <perkj@chromium.org> |
Support texture scaling in Androids MediaEncoder.
This cl make it possible for the hw video encoder to downscale a texture image before encoding. The purpose is to allow downscaling if the quality is too bad at the current resolution.
BUG=webrtc:4993
R=magjed@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1470043002 .
Cr-Commit-Position: refs/heads/master@{#10804}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| a3c20bb9a096495c5f8a876329b5edcedcf04ab8 |
26-Nov-2015 |
Per <perkj@chromium.org> |
Add support for scaling textures in AndroidVideoCapturer.
The idea is to also reuse AndroidTextureBuffer::CropAndScale when scaling in the encoder.
BUG=webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1471333003 .
Cr-Commit-Position: refs/heads/master@{#10802}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
|
| fac0655fd7fe0b40ef50dc5b7f11ea44d72cec6c |
25-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding the ability to create an RtpSender without a track.
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
pp/webrtc/audiotrack.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/videotrack.cc
pp/webrtc/webrtcsession.cc
|
| 444682acf9804c5fcbddaded9e900ba3cc6921fc |
25-Nov-2015 |
qiangchen <qiangchen@chromium.org> |
Remove frame time scheduing in IncomingVideoStream
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| b2514725a9a2e09da15f77a2ab9a6446a4a616f7 |
24-Nov-2015 |
ivoc <ivoc@webrtc.org> |
Add JNI interface for functions to start and stop recording AEC dumps and RTC event logs.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1409323009
Cr-Commit-Position: refs/heads/master@{#10776}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 4c5eea3c73e90b11bc17679d6b0943813e4c5038 |
24-Nov-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceViewRenderer: Don't rely on widthSpec/heightSpec after onMeasure() returns
SurfaceViewRenderer currently stores widthSpec/heightSpec internally, and triggers requestLayout() from renderFrameOnRenderThread()->checkConsistentLayout() when it detects a change using widthSpec/heightSpec. This is not reliable, because onMeasure() might be called several times during the layout process negotiation. For example it might look like this:
-> onMeasure(at most 1920, at most 1080)
<- setMeasuredDimension(1080, 1080)
-> onMeasure(exactly 1080, exactly 1080)
<- setMeasuredDimension(1080, 1080)
Then we store (exactly 1080, exactly 1080) even though we are allowed to be bigger than this, and requestLayout() will never be triggered.
This CL moves the requestLayout() trigger to updateFrameDimensionsAndReportEvents() when the frame size changes.
Other small changes in this CL are:
* Replace with/height variables with Point.
* Add logging in updateFrameDimensionsAndReportEvents() even when rendererEvents is null.
* Use Math.round() in RendererCommon.getDisplaySize() instead of integer cast.
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1453413005 .
Cr-Commit-Position: refs/heads/master@{#10774}
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 7baf79fb9ece918f8ad4768725529af5a367e0d7 |
24-Nov-2015 |
perkj <perkj@webrtc.org> |
Temporary remove spamming MediaDecoder log
This log will write for each decoded frame if the textures are rendered using VideoRenderGUI
and the the screen is locked.
TBR=glaznew@webrtc.org
Review URL: https://codereview.webrtc.org/1465093004
Cr-Commit-Position: refs/heads/master@{#10771}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 4f2152e3286b03292836fd95813cc95933c79c4d |
24-Nov-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceViewRenderer: Make sure not to call eglCreateSurface() twice
eglCreateSurface() calls are posted to the render thread from both init() and surfaceCreated(). If the render thread does not process the eglCreateSurface() message from init() before surfaceCreated() is called, eglCreateSurface() will be called twice resulting in a crash.
This CL makes sure eglCreateSurface() is only called once.
BUG=b/25815604
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1466133002 .
Cr-Commit-Position: refs/heads/master@{#10769}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 9237559b16841c6a7762cfe015e83fb7dd3652e6 |
24-Nov-2015 |
perkj <perkj@webrtc.org> |
Add SurfaceTextureHelper.disconnect(Handler handler) method
This method should be used when the SurfaceTextureHelper is created to use a specific handler.
This now guarantee that the looper used by handler is destroyed after a frame has been returned.
Review URL: https://codereview.webrtc.org/1465163003
Cr-Commit-Position: refs/heads/master@{#10767}
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| b5cb19b37c361a263a9cec2e2fb356d16520afd1 |
24-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing direction attribute in answer for non-RTP protocols.
"non-RTP protocols" refers to SCTP data channels. Because
there are no streams for SCTP data channels, the answer was being
set to RECVONLY.
BUG=webrtc:5228
Review URL: https://codereview.webrtc.org/1473013002
Cr-Commit-Position: refs/heads/master@{#10762}
pp/webrtc/peerconnection_unittest.cc
ession/media/mediasession.cc
|
| 05816eb8d7ec4fe6877339ba5b9fc6412364e436 |
24-Nov-2015 |
wr.wllm <wr.wllm@gmail.com> |
Fix target_arch for ios devices
Replace armv7 by arm and arm64 in documentation for iOS build
instructions.
BUG=5125
Review URL: https://codereview.webrtc.org/1418513014
Cr-Commit-Position: refs/heads/master@{#10761}
pp/webrtc/objc/README
|
| 1aa6efe885a130da8272542309b70116497104a7 |
23-Nov-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android ThreadUtils: Make the class public for access outside org.webrtc
Also make the class non-final. We shouldn't use non-final classes, because then we can't mock them.
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1470973002 .
Cr-Commit-Position: refs/heads/master@{#10757}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
|
| 8becec3b4960883a3032c2b15056ae4678132198 |
23-Nov-2015 |
tfarina <tfarina@chromium.org> |
talk: remove deprecated *processor.h files
Chromium's libjingle gyp/gn files has been updated already.
BUG=None
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1458133004
Cr-Commit-Position: refs/heads/master@{#10745}
edia/base/fakemediaprocessor.h
edia/base/mediaengine.h
edia/base/voiceprocessor.h
|
| 87d584597c99195758b0f115f776a18f15f32ffb |
23-Nov-2015 |
perkj <perkj@webrtc.org> |
Fix androidmediadecoder_jni TS logging.
And fix pragma warning about deprecated "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h include.
Review URL: https://codereview.webrtc.org/1461273002
Cr-Commit-Position: refs/heads/master@{#10744}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
|
| 43edf0ffb91a50e2efa01c7befe4d188a7e30ea2 |
21-Nov-2015 |
stefan <stefan@webrtc.org> |
Require negotiation to send transport cc feedback over RTCP.
BUG=4312
Review URL: https://codereview.webrtc.org/1452883002
Cr-Commit-Position: refs/heads/master@{#10735}
edia/base/codec.cc
edia/base/codec.h
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| bd13838ccc87f94d1e951bcf780979622b020359 |
21-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1457653003
Cr-Commit-Position: refs/heads/master@{#10734}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 5def7b9fdea0d027bca3df734d86fb877a83bdbf |
20-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
Reason for revert:
Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection.
Original issue's description:
> Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
>
> Reason for revert:
> Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
>
> Original issue's description:
> > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
> >
> > Reason for revert:
> > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
> >
> > Original issue's description:
> > > Adding the ability to create an RtpSender without a track.
> > >
> > > This CL also changes AddStream to immediately create a sender, rather
> > > than waiting until the track is seen in SDP. And the PeerConnection now
> > > builds the list of "send streams" from the list of senders, rather than
> > > the collection of local media streams.
> > >
> > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > > Cr-Commit-Position: refs/heads/master@{#10414}
> >
> > TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> >
> > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> > Cr-Commit-Position: refs/heads/master@{#10417}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae
> Cr-Commit-Position: refs/heads/master@{#10730}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1460323002
Cr-Commit-Position: refs/heads/master@{#10732}
pp/webrtc/audiotrack.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/videotrack.cc
pp/webrtc/webrtcsession.cc
|
| 7add0584390dcfb236165a6472ede6c2a94eaeed |
20-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Move some receive stream configuration into webrtc::AudioReceiveStream.
Simplify creation of VoE channels and Call streams in WVoMC.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1454073002
Cr-Commit-Position: refs/heads/master@{#10731}
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 6834fa10f142bf5e2275142acb834898911d09ae |
20-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
Reason for revert:
Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
Original issue's description:
> Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
>
> Reason for revert:
> Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
>
> Original issue's description:
> > Adding the ability to create an RtpSender without a track.
> >
> > This CL also changes AddStream to immediately create a sender, rather
> > than waiting until the track is seen in SDP. And the PeerConnection now
> > builds the list of "send streams" from the list of senders, rather than
> > the collection of local media streams.
> >
> > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > Cr-Commit-Position: refs/heads/master@{#10414}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> Cr-Commit-Position: refs/heads/master@{#10417}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1413983004
Cr-Commit-Position: refs/heads/master@{#10730}
pp/webrtc/audiotrack.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/videotrack.cc
pp/webrtc/webrtcsession.cc
|
| 30e918278c8e0221ebbb24727fca90676da77220 |
20-Nov-2015 |
perkj <perkj@webrtc.org> |
This cl add support to encode from textures to MediaCodecVideoEncoder.
This has also partly been reviewed in https://codereview.webrtc.org/1375953002/.
BUG=webrtc:4993
TBR=glaznew@webrtc.org
Review URL: https://codereview.webrtc.org/1403713002
Cr-Commit-Position: refs/heads/master@{#10725}
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 7e63ef0e8f3baf832005e2e378b6834c0d005f12 |
20-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Allow default audio receive channel to receive on any unsignalled SSRC.
BUG=webrtc:5208
Review URL: https://codereview.webrtc.org/1455923003
Cr-Commit-Position: refs/heads/master@{#10723}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 17c0aff9eab8f2977c3ced1c9248cc10810f087e |
20-Nov-2015 |
Alex Glaznev <glaznev@google.com> |
Enable VP9 HW decoder on Exynos chips.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1466543002 .
Cr-Commit-Position: refs/heads/master@{#10720}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 7755e2064b5b8add2ff0c9d0b5d3fb34ee1726d1 |
19-Nov-2015 |
perkj <perkj@webrtc.org> |
Chrome has now been updated.
CapturedFrame
Removed deprecated elapsed_time.
Changed rotation to be webrtc::VideoRotation.
WebRTCVideoFrame
Removed deprecated InitToBlack
Removed deprecated constructors.
Review URL: https://codereview.webrtc.org/1461053002
Cr-Commit-Position: refs/heads/master@{#10718}
pp/webrtc/androidvideocapturer.cc
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoframefactory.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 726b1f7a1467a33b1c3feedff84fca953f7f9c1d |
19-Nov-2015 |
perkj <perkj@webrtc.org> |
Removed dummy "mediastreamsignaling.h"
Review URL: https://codereview.webrtc.org/1460483005
Cr-Commit-Position: refs/heads/master@{#10717}
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
|
| 191c1f9d5bc1a4adbcaf87fe93214c54b3530dc8 |
19-Nov-2015 |
ivoc <ivoc@webrtc.org> |
Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.
NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:5231
Review URL: https://codereview.webrtc.org/1462933002
Cr-Commit-Position: refs/heads/master@{#10716}
pp/webrtc/peerconnection_unittest.cc
|
| ef453238aaf9d4b5c7ded9519c571e56205b0978 |
19-Nov-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Make classes non-final
The classes are not mockable if they are final.
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1459873002 .
Cr-Commit-Position: refs/heads/master@{#10714}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
|
| 1503867850670447624c8227aea26b038454295b |
19-Nov-2015 |
ivoc <ivoc@webrtc.org> |
Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.
NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:5231
Review URL: https://codereview.webrtc.org/1459883002
Cr-Commit-Position: refs/heads/master@{#10710}
pp/webrtc/peerconnection_unittest.cc
|
| b6755ab6df946af2ceabd657f559b817276141df |
19-Nov-2015 |
henrika <henrika@webrtc.org> |
Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ )
Reason for revert:
Reverting since this fix might hide real issue and the reported root problem seems extremely rare.
Original issue's description:
> Adding thread timeout for audio recorer thread in Java
>
> BUG=NONE
>
> Committed: https://crrev.com/fd614c2149c7985bd83df809df71d0d60e5a8f74
> Cr-Commit-Position: refs/heads/master@{#10671}
TBR=magjed@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE
Review URL: https://codereview.webrtc.org/1459123002
Cr-Commit-Position: refs/heads/master@{#10707}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
ibjingle.gyp
|
| 488e75f11b840dfbe636a9ea9bbc18252e7c59f0 |
19-Nov-2015 |
Per <perkj@chromium.org> |
Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/
It do the following:
The SurfaceTexture.updateTexImage() calls are moved from the video renderers into MediaCodecVideoDecoder, and the destructor of the texture frames will signal MediaCodecVideoDecoder that the frame has returned. This CL also removes the SurfaceTexture from the native handle and only exposes the texture matrix instead, because only the video source should access the SurfaceTexture.
It moves the responsibility of calculating the decode time to Java.
Patchset2 Refactor MediaCodecVideoDecoder to drop frames if a texture is not released.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440343002 .
Cr-Commit-Position: refs/heads/master@{#10706}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| 521ed7bf022c4e30574d7970c2be5be46567f4cd |
19-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Convert internal representation of Srtp cryptos from string to int
TBR=pthatcher@webrtc.org
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1458023002 .
Cr-Commit-Position: refs/heads/master@{#10703}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
edia/base/cryptoparams.h
ession/media/channel.cc
ession/media/channel.h
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 318166bed75dcbc00a7b79f715f9953aff9ffbc7 |
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
Reason for revert:
Broke chromium fyi build.
Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}
TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1455233005
Cr-Commit-Position: refs/heads/master@{#10702}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
edia/base/cryptoparams.h
ession/media/channel.cc
ession/media/channel.h
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 2764e1027a08a5543e04b854a27a520801faf6eb |
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Convert internal representation of Srtp cryptos from string to int.
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.
External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1416673006
Cr-Commit-Position: refs/heads/master@{#10701}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
edia/base/cryptoparams.h
ession/media/channel.cc
ession/media/channel.h
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| b7ce96470b99510937e489bcb4dc3165a9ab1b28 |
18-Nov-2015 |
kjellander@webrtc.org <kjellander@google.com> |
modules/video_coding/utility: Remove include
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440873005 .
Cr-Commit-Position: refs/heads/master@{#10699}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| ad948c42a1fd29bf22205ded2a175a967748abe4 |
18-Nov-2015 |
Alex Glaznev <glaznev@google.com> |
Preliminary support of VP9 HW encoder on Android.
Not fully tested yet. Verified in test loopback application
with fake VP9 codec factory.
Assume that encoder generates bitstream in non flexible mode with
one temporal and one spatial layers.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1451953002 .
Cr-Commit-Position: refs/heads/master@{#10695}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 2557b86e7648ffebc5781df9f093ca5a84efc219 |
18-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
modules/video_coding refactorings
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
edia/webrtc/fakewebrtcvideoengine.h
|
| 4dd7a653b5011495f4c805461bad33405c1fb1b8 |
18-Nov-2015 |
phoglund <phoglund@webrtc.org> |
Temporarily disable VERIFY while bug is investigated.
This breaks some client apps in annoying ways, so disable for now.
BUG=webrtc:4776
Review URL: https://codereview.webrtc.org/1461513003
Cr-Commit-Position: refs/heads/master@{#10693}
pp/webrtc/videosource.cc
|
| 2aff615bd7c7c24a6e7a35163112f169ff4f9246 |
18-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Remove spammy logging of RTCP delivery failures.
Since BundleFilter doesn't filter RTCP anymore we can have incoming
RTCPs for audio delivered to video, that delivery will fail when there
are no video receivers causing the log to be spammed.
BUG=webrtc:5223
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1458853002 .
Cr-Commit-Position: refs/heads/master@{#10687}
edia/webrtc/webrtcvideoengine2.cc
|
| fd614c2149c7985bd83df809df71d0d60e5a8f74 |
17-Nov-2015 |
henrika <henrika@webrtc.org> |
Adding thread timeout for audio recorer thread in Java
BUG=NONE
Review URL: https://codereview.webrtc.org/1444313002
Cr-Commit-Position: refs/heads/master@{#10671}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
ibjingle.gyp
|
| 6f8ce060a21fcdc1c951fbf06768eb0cc0083b2f |
16-Nov-2015 |
kjellander <kjellander@webrtc.org> |
common_video: rename interface -> include
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1418913006
Cr-Commit-Position: refs/heads/master@{#10659}
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/surfacetexturehelper_jni.h
edia/base/videoframe.h
edia/webrtc/webrtcvideoframe.h
|
| 482b12e2c3fedfe94a7c3fd665cbe77b848f1b31 |
16-Nov-2015 |
pbos <pbos@webrtc.org> |
Remove BundleFilter filtering of RTCP.
BundleFilter may not know the remote SSRC for all incoming RTCP packets,
so there's no point in filtering them.
BUG=webrtc:4740
R=hta@webrtc.org, juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1437683005
Cr-Commit-Position: refs/heads/master@{#10655}
ession/media/bundlefilter.cc
ession/media/bundlefilter.h
ession/media/bundlefilter_unittest.cc
ession/media/channel.cc
ession/media/channel_unittest.cc
|
| 3a94154035fa16e4efd91125311f076b547c38b9 |
16-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Move some send stream configuration into webrtc::AudioSendStream.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1418503010
Cr-Commit-Position: refs/heads/master@{#10652}
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 633a3aa26fbe9dc40df880ba4ffa7b863f11e473 |
16-Nov-2015 |
magjed <magjed@webrtc.org> |
ThreadUtils: Add joinUninterruptibly() with timeout
This is similar to com.google.common.util.concurrent.Uninterruptibles.joinUninterruptibly().
http://docs.guava-libraries.googlecode.com/git/javadoc/com/google/common/util/concurrent/Uninterruptibles.html#joinUninterruptibly(java.lang.Thread,%20long,%20java.util.concurrent.TimeUnit)
Review URL: https://codereview.webrtc.org/1444273002
Cr-Commit-Position: refs/heads/master@{#10651}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
|
| 3e0f602055f2980d805b0f9b4aa584e675788925 |
16-Nov-2015 |
magjed <magjed@webrtc.org> |
Android EglBase: Add support for creating EGLSurface from Surface, not SurfaceHolder
Review URL: https://codereview.webrtc.org/1438223003
Cr-Commit-Position: refs/heads/master@{#10646}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 4a41361f9819b995946a87e32abe008734528cee |
13-Nov-2015 |
magjed <magjed@webrtc.org> |
Android SurfaceViewRenderer: Never hold a pending frame indefinitely
The original purpose with keeping one pending frame in SurfaceViewRenderer was to reduce latency for the first rendered frame when we are waiting for the Surface to be created. However, it is very dangerous to hold a pending frame indefinitely when used with a SurfaceTexture, because the SurfaceTexture only has one frame and thus holding a frame in the renderer will freeze everything and typically cause timeout crashes.
Review URL: https://codereview.webrtc.org/1435413006
Cr-Commit-Position: refs/heads/master@{#10638}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| c01c25434ba92f6ea32cdfdcde77ec8278182851 |
13-Nov-2015 |
Per <perkj@chromium.org> |
Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ )
Reason for revert:
Causes fallback to SW decoder if a renderer is put in the background.
Original issue's description:
> Patchset 1 is a pure
> revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
>
> Following patchsets move the responsibility of calculating the decode time to Java.
>
> TESTED= Apprtc loopback using H264 and VP8 on N5, N6, N7, S5
>
> Committed: https://crrev.com/9cb8982e64f08d3d630bf7c3d2bcc78c10db88e2
> Cr-Commit-Position: refs/heads/master@{#10597}
TBR=magjed@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
Review URL: https://codereview.webrtc.org/1441363002 .
Cr-Commit-Position: refs/heads/master@{#10637}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| cbe9f51cf85a5aeb20a5134dad56cd2b527c098d |
13-Nov-2015 |
phoglund <phoglund@webrtc.org> |
Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ )
Reason for revert:
Unfortunately this breaks an internal downstream project since we have an ancient libsrtp. Reverting until we can figure out how to update our libsrtp.
Original issue's description:
> Remove global list of SRTP sessions.
> Instead save a reference to the SrtpSession inside the srtp_ctx_t.
>
> BUG=webrtc:5133
>
> Committed: https://crrev.com/9cafd972779ed7b25886ab276e0ede7b7a8b76a1
> Cr-Commit-Position: refs/heads/master@{#10591}
TBR=juberti@google.com,juberti@webrtc.org,jbauch@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5133
Review URL: https://codereview.webrtc.org/1442863003
Cr-Commit-Position: refs/heads/master@{#10635}
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
|
| faac497af560ece34301343eb40377fd5503f7a0 |
13-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Fix for scenario where m-line is revived after being set to port 0.
When this is detected, we'll now "reconfigure" the senders and
receivers, which will reconnect the capturers/renderers to the
underlying streams which have been recreated.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1428243005
Cr-Commit-Position: refs/heads/master@{#10628}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
|
| 68876f990ea1ea365d2d8155df261b38ec9fbeff |
12-Nov-2015 |
Patrik Höglund <phoglund@webrtc.org> |
Introduces Android API level linting, fixes all current API lint errors.
This CL attempts to annotate accesses on >16 API levels using as
small scopes as possible. The TargetApi notations mean "yes, I know
I'm accessing a higher API and I take responsibility for gating the
call on Android API level". The Encoder/Decoder classes are annotated
on the whole class, but they're only accessed through JNI; we should
annotate on method level otherwise and preferably on private methods.
This patch also fixes some compiler-level deprecation warnings (i.e.
-Xlint:deprecation), but probably not all of them.
BUG=webrtc:5063
R=henrika@webrtc.org, kjellander@webrtc.org, magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1412673008 .
Cr-Commit-Position: refs/heads/master@{#10624}
pp/webrtc/java/android/org/webrtc/Camera2Enumerator.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/CameraEnumerator.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
ibjingle.gyp
|
| 9576e548368b34e150c3e6d19e889de9f0f67e96 |
12-Nov-2015 |
perkj <perkj@webrtc.org> |
Reland "Prepare MediaCodecVideoEncoder for surface textures.""
This reverts commit 12f680214e28dc5f0a13ac8afc0d1445f89e67e6.
Original cl in https://codereview.webrtc.org/1396073003/
Prepare MediaCodecVideoEncoder for surface textures.
This refactors MediaVideoEncoder to prepare for adding support to encode from textures. The C++ layer does not have any functional changes.
- Moves ResetEncoder to always work on the codec thread
- Adds use of ThreadChecker.
- Change Java MediaEncoder.Init to return true or false and introduce method getInputBuffers.
- Add simple unit test for Java MediaCodecVideoEncoder.
The pure revert of the revert is in patchset 1.
Patchset 2, moves getting the input buffer to before storing pending timestamps etc to fix b/24984012.
BUG=webrtc:4993 b/24984012
Review URL: https://codereview.webrtc.org/1406203002
Cr-Commit-Position: refs/heads/master@{#10622}
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 8093d5442e4c365bfebc07abcf5fb653bd7a1d57 |
12-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Change default SSRC for RTCP receiver reports to not collide with video.
BUG=chromium:547661
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1438183002
Cr-Commit-Position: refs/heads/master@{#10621}
edia/webrtc/webrtcvoiceengine.h
|
| 5dda80abea311731144b1d544aff61c408412f12 |
12-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
Remove webrtc/modules/video_{capture,render}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=pbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1439823002 .
Cr-Commit-Position: refs/heads/master@{#10619}
edia/devices/mobiledevicemanager.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
|
| fc6affc60d2298600c1c8433fd918226f4f38a5b |
12-Nov-2015 |
magjed <magjed@webrtc.org> |
Android SurfaceViewRenderer: Call glClear() for every frame to avoid bad GL state
BUG=webrtc:5147
Review URL: https://codereview.webrtc.org/1436883002
Cr-Commit-Position: refs/heads/master@{#10617}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 653b8e02f22c9b6ba38be1cf4c0fa101894a9407 |
11-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
Reason for revert:
Relanding with compile warning fixed.
Original issue's description:
> Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
>
> Reason for revert:
> Caused compiler warning, breaking Chrome FYI bots.
>
> Original issue's description:
> > Adding the ability to change ICE servers through SetConfiguration.
> >
> > Added a SetIceServers method to PortAllocator. Also added a new
> > PeerConnection Initialize method that takes a PortAllocator, in the
> > hope that we can get rid of PortAllocatorFactoryInterface, since the
> > only substantial thing a factory does is convert the webrtc:: ICE
> > servers to cricket:: versions.
> >
> > Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> > Cr-Commit-Position: refs/heads/master@{#10420}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/18a944bf0ac9eed872dc009bd58e6bc12c946303
> Cr-Commit-Position: refs/heads/master@{#10421}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1414313003
Cr-Commit-Position: refs/heads/master@{#10609}
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
|
| 718b6c72ae3c8a88ff4bbe672874e122a7c15bff |
11-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Add waiting to SetSendSsrc tests.
These tests were flaky since a paced packet could arrive between
consecutive calls to NumRtpPackets or NumRtpBytes.
BUG=webrtc:5193
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1436753003 .
Cr-Commit-Position: refs/heads/master@{#10603}
edia/base/videoengine_unittest.h
|
| fa566d610fe26a7007246ad84295aaaac6da15cf |
11-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Remove webrtc/examples/android/media_demo.
The JNI code for VoiceEngine is not maintained and VoiceEngine is being
refactored. This is not a supported Java interface, use AppRTCDemo as a
starting point instead.
Also renames webrtc/libjingle_examples.gyp webrtc/webrtc_examples.gyp to
replace the previous file (that only contained media_demo).
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1439593002 .
Cr-Commit-Position: refs/heads/master@{#10599}
ibjingle_tests.gyp
|
| cbfabbf818717bfaea1740779cf199b740a69e5f |
11-Nov-2015 |
perkj <perkj@webrtc.org> |
Fix potential tearing issue in VideoRendererGui.
This make sure that the texture copy is syncronized.
To reproduce the problem I:
Reverted "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/"
commit 543b6ca30a43eeb069c699291460ce6bacc7959d.
Reverted "Enable SurfaceViewRenderer for AppRTCDemo"
commit 7076729c57c27aa813760d2038be02c36f4d7649.
and ran ApprtDemo in loopback and changed the orientation a couple of times.
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1437823002
Cr-Commit-Position: refs/heads/master@{#10598}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 9cb8982e64f08d3d630bf7c3d2bcc78c10db88e2 |
11-Nov-2015 |
perkj <perkj@webrtc.org> |
Patchset 1 is a pure
revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
Following patchsets move the responsibility of calculating the decode time to Java.
TESTED= Apprtc loopback using H264 and VP8 on N5, N6, N7, S5
Review URL: https://codereview.webrtc.org/1422963003
Cr-Commit-Position: refs/heads/master@{#10597}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| b2d1c5026dc3486670d2ffc7f663be3265bf18b9 |
11-Nov-2015 |
magjed <magjed@webrtc.org> |
SurfaceViewRenderer: Add resource name to log outputs and exceptions
Add resource name to log outputs to distinguish local renderer from remote renderer.
This Cl also adds some thread checks and factors out a small helper function makeBlack().
Review URL: https://codereview.webrtc.org/1420203003
Cr-Commit-Position: refs/heads/master@{#10596}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 5237aaf243d29732f59557361b7a993c0a18cf0e |
11-Nov-2015 |
tfarina <tfarina@chromium.org> |
Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/peerconnection.cc
pp/webrtc/videosource.cc
pp/webrtc/webrtcsdp.cc
edia/base/capturemanager_unittest.cc
edia/base/streamparams_unittest.cc
edia/base/testutils.cc
edia/base/testutils.h
edia/base/videocommon.cc
edia/base/videoframe.cc
edia/devices/devicemanager_unittest.cc
edia/devices/win32devicemanager.cc
edia/sctp/sctpdataengine.cc
edia/webrtc/simulcast.cc
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel_unittest.cc
|
| 9cafd972779ed7b25886ab276e0ede7b7a8b76a1 |
10-Nov-2015 |
jbauch <jbauch@webrtc.org> |
Remove global list of SRTP sessions.
Instead save a reference to the SrtpSession inside the srtp_ctx_t.
BUG=webrtc:5133
Review URL: https://codereview.webrtc.org/1416093010
Cr-Commit-Position: refs/heads/master@{#10591}
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
|
| 9af97f89103d8f1f77b52a6ae77b8b7bcdc23f71 |
10-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
WebRTC should generate default private address even when adapter enumeration is disabled.
Introduce a DefaultAddressProvider such that rtc::Network can't access other part of NetworkManager.
This also removes the hack of generating the loopback address. The dependency has been removed by https://codereview.chromium.org/1417023003/
BUG=webrtc:5061
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1411253008 .
Cr-Commit-Position: refs/heads/master@{#10590}
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| be57983f4bd875c39a229bab5112b32dad004057 |
10-Nov-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Rename Maybe to Optional
And add examples of good and bad usage to the documentation.
R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1432553007 .
Cr-Commit-Position: refs/heads/master@{#10588}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/videosource_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
|
| 69a7fd50476a60ec3def8552993bebef83ed9c58 |
10-Nov-2015 |
Alex Glaznev <glaznev@google.com> |
Support VP9 HW video decoding on Android.
Preliminary verification is done for OMX.google.vp9.decoder codec.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1416023005 .
Cr-Commit-Position: refs/heads/master@{#10586}
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 3ed348707e93980fd74246f7a1dfab011f841087 |
10-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Remove field trial check for VP9.
VP9 is put as second codec in supported codec list.
BUG=chromium:500602
Review URL: https://codereview.webrtc.org/1432673002
Cr-Commit-Position: refs/heads/master@{#10577}
edia/webrtc/webrtcvideoengine2.cc
|
| ce83ae1c19eb3fb8aea84d8e02c2c005115e0440 |
10-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Improve informative message in codereview.settings.
In https://codereview.webrtc.org/1389963002 the message
displayed when trying to create a CL from an unsupported
location was improved. However it's confusing for developers
working from a WebRTC checkout if they stand in src/webrtc
when trying to create a CL.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1432073002 .
Cr-Commit-Position: refs/heads/master@{#10571}
odereview.settings
|
| 83dfad685322119ffe85ff5670cabbbaf385a111 |
09-Nov-2015 |
perkj <perkj@webrtc.org> |
VideoCapturerAndroid: Changed camera freeze check to check that all frames are pending before reporting a client error.
BUG=b/25514149
Review URL: https://codereview.webrtc.org/1423073006
Cr-Commit-Position: refs/heads/master@{#10563}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 89ef6cc13e8aa9f16b212dd82124f4297a1f7385 |
09-Nov-2015 |
perkj <perkj@webrtc.org> |
Attempt to open Android camera later if it is already in use.
This change VideoCapturerAndroid to attempt 3 times with a period of 300ms to open the camera if it fails.
This is so that if another application have it already opened, it would have more time to release it.
BUG=b/25190234
Review URL: https://codereview.webrtc.org/1422023007
Cr-Commit-Position: refs/heads/master@{#10559}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 1ebf8ba3681b33b582766bceef922563614c1d47 |
09-Nov-2015 |
magjed <magjed@webrtc.org> |
SurfaceViewRenderer: Drop old frames instead of new frames
If SurfaceViewRenderer can't keep up with the stream of incoming frames it has to drop frames. Currently, new frames are dropped until the old pending frame is rendered. This CL drops the old pending frame instead.
Review URL: https://codereview.webrtc.org/1417063005
Cr-Commit-Position: refs/heads/master@{#10558}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 3bfef44a4de22c562bdfd787872ef4a13aa1ad60 |
08-Nov-2015 |
perkj <perkj@webrtc.org> |
Changed timeout to 6s for reporting android camera freeze.
Also distinguish between camera failures and failures due to that buffers has not been returned.
Adds unit tests for making sure CameraEventHandler.onError is triggered if frames are not returned.
BUG=b/25514149
Review URL: https://codereview.webrtc.org/1415013006
Cr-Commit-Position: refs/heads/master@{#10555}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 566ef247b9779f6c9d0e7ec9eea6b037f4682c53 |
07-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1403363003
Cr-Commit-Position: refs/heads/master@{#10548}
pp/webrtc/mediacontroller.cc
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 23725e09c6533739afa67ca9694f6fa874c26279 |
06-Nov-2015 |
noahric <noahric@chromium.org> |
Remove ICU usage from jni_helpers.cc.
JNI already has jstring<->UTF8 string conversion, so using that should
save ~1mb off android binaries (ICU is *large*), probably around
300-400k after compression.
BUG=
Review URL: https://codereview.webrtc.org/1430023005
Cr-Commit-Position: refs/heads/master@{#10545}
pp/webrtc/java/jni/jni_helpers.cc
uild/common.gypi
ibjingle.gyp
|
| 0ccae135562ac180da053fcecda91a0365621f14 |
03-Nov-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Changed FakeVoiceEngine into a MockVoiceEngine.
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1402403008 .
Cr-Commit-Position: refs/heads/master@{#10491}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 5a846c086bdf714c9b8ca55bb1e93b04517cc06a |
02-Nov-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Make ConnectionType public in order to add java NetworkObserver.
BUG=
R=glaznev@webrtc.org, jiayl@google.com
Review URL: https://codereview.webrtc.org/1429053002 .
Cr-Commit-Position: refs/heads/master@{#10485}
pp/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java
|
| 5c3da4b6e9bdc787a3c1f25c0ff0c16a4271fd26 |
30-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Call MediaCodec.stop() on separate thread.
MediaCodec.stop() call may hang in some rear cases. To avoid
application hang this call need to be done on separate thread and
possible error reported back to application.
Application may elect to continue executing and use another codec
instance for encoding/decoding or stop the call and exit.
BUG=b/24339249
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1425143005 .
Cr-Commit-Position: refs/heads/master@{#10467}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 102c6a61bc0b42dc0956d013530fc0213b7e881b |
30-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
Replace rtc::cricket::Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).
This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.
Review URL: https://codereview.webrtc.org/1430433004
Cr-Commit-Position: refs/heads/master@{#10461}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/videosource_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
|
| bbaf3633c54e3d49aa4c762b8eaa34e09de01c45 |
29-Oct-2015 |
Stefan Holmer <stefan@webrtc.org> |
Filter overlapping RTP header extensions.
This removes unnecessary RTP header extension overhead since only one of
these extensions is used at a time.
BUG=webrtc:4254
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1429753003 .
Cr-Commit-Position: refs/heads/master@{#10455}
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 075fb4bfea8f3fc487b63e96f124b2dfd21b7f92 |
29-Oct-2015 |
asapersson <asapersson@webrtc.org> |
MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
BUG=
Review URL: https://codereview.webrtc.org/1426033002
Cr-Commit-Position: refs/heads/master@{#10453}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| e55c42c13ee2620b46376ab708e7d4c0d698cf51 |
28-Oct-2015 |
glaznev <glaznev@webrtc.org> |
Remove limitation on the amount of maximum pending HW decoder inputs.
Plus log first few decoder frames in and out events.
BUG=b/25287910
Review URL: https://codereview.webrtc.org/1423843005
Cr-Commit-Position: refs/heads/master@{#10439}
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnection.cc
edia/devices/yuvframescapturer.cc
edia/webrtc/simulcast.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvoiceengine.cc
|
| 83585c9075a3f0dad24a55b24872168bf825eacc |
28-Oct-2015 |
magjed <magjed@webrtc.org> |
VideoCapturerAndroid: More frequent and verbose logging
BUG=b/24437529
Review URL: https://codereview.webrtc.org/1417633007
Cr-Commit-Position: refs/heads/master@{#10434}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| ec9d187f708933c75c3b48cf62296c37c7c506d9 |
27-Oct-2015 |
rlester <rlester@google.com> |
Added override keyword to overridden methods to stop compiler warnings.
BUG=
Review URL: https://codereview.webrtc.org/1417543002
Cr-Commit-Position: refs/heads/master@{#10433}
edia/webrtc/webrtcvideoframe.h
ession/media/channel.h
|
| 27f6fd346accd7a96ff2efde4af3a80509b1b92f |
27-Oct-2015 |
pbos <pbos@webrtc.org> |
Remove noparent from talk/OWNERS.
Lets webrtc root OWNERS approve talk/ code as well.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1413773005
Cr-Commit-Position: refs/heads/master@{#10427}
WNERS
|
| 85a0496b8c4ac01da7c716ea7950093659864c8e |
27-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Implement AudioSendStream::GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1414743004
Cr-Commit-Position: refs/heads/master@{#10424}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 18a944bf0ac9eed872dc009bd58e6bc12c946303 |
27-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
Reason for revert:
Caused compiler warning, breaking Chrome FYI bots.
Original issue's description:
> Adding the ability to change ICE servers through SetConfiguration.
>
> Added a SetIceServers method to PortAllocator. Also added a new
> PeerConnection Initialize method that takes a PortAllocator, in the
> hope that we can get rid of PortAllocatorFactoryInterface, since the
> only substantial thing a factory does is convert the webrtc:: ICE
> servers to cricket:: versions.
>
> Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> Cr-Commit-Position: refs/heads/master@{#10420}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1424803004
Cr-Commit-Position: refs/heads/master@{#10421}
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
|
| d3b26d94399ff539db375a9b84010ee75479d4cf |
27-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding the ability to change ICE servers through SetConfiguration.
Added a SetIceServers method to PortAllocator. Also added a new
PeerConnection Initialize method that takes a PortAllocator, in the
hope that we can get rid of PortAllocatorFactoryInterface, since the
only substantial thing a factory does is convert the webrtc:: ICE
servers to cricket:: versions.
Review URL: https://codereview.webrtc.org/1391013007
Cr-Commit-Position: refs/heads/master@{#10420}
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
|
| 8f46c63f6f764254892f4111b54aa1cc8f32eeeb |
26-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Reason for revert:
Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1426443007
Cr-Commit-Position: refs/heads/master@{#10417}
pp/webrtc/audiotrack.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/videotrack.cc
pp/webrtc/webrtcsession.cc
ibjingle_tests.gyp
|
| ac9d92ccbe2b29590c53f702e11dc625820480d5 |
26-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding the ability to create an RtpSender without a track.
This CL also changes AddStream to immediately create a sender, rather
than waiting until the track is seen in SDP. And the PeerConnection now
builds the list of "send streams" from the list of senders, rather than
the collection of local media streams.
Review URL: https://codereview.webrtc.org/1413713003
Cr-Commit-Position: refs/heads/master@{#10414}
pp/webrtc/audiotrack.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/videotrack.cc
pp/webrtc/webrtcsession.cc
ibjingle_tests.gyp
|
| 4cba4eba596706f2238d14f96f4e181f47e5034c |
26-Oct-2015 |
pbos <pbos@webrtc.org> |
Disable denoising for VP9 by default.
BUG=webrtc:5108
R=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1418133012
Cr-Commit-Position: refs/heads/master@{#10413}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 5d9b92b53daf4db78fd090be4e210e07f786120d |
24-Oct-2015 |
noahric <noahric@chromium.org> |
Update Bind to match its comments and always capture by value. Also update the generated count to 9 args.
The existing comment is wrong, and the test even ensures it: Bind will capture reference values by reference. That makes it hard to use with AsyncInvoker, because you can't safely Bind to a function that takes (const) reference params.
The new version of this code strips references in the bound object, so it captures by value, but can bind against functions that take const references, they'll just be references to the copy.
As the class comment implies, actual by-reference args should be passed as pointers or things that safely share (e.g. scoped_refptr) and not references directly. A new test case ensures the pointer reference works. The new code will also give a compiler error if you try to bind
to a non-const reference.
BUG=
Review URL: https://codereview.webrtc.org/1291543006
Cr-Commit-Position: refs/heads/master@{#10397}
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
|
| 401fb0648a262b9fe3981f9e2957cbca6682cb9b |
24-Oct-2015 |
magjed <magjed@webrtc.org> |
SurfaceTextureHelper: Remove use of quitSafely() because it's API lvl 18
There is no reason to not just quit() in release().
Review URL: https://codereview.webrtc.org/1418563005
Cr-Commit-Position: refs/heads/master@{#10394}
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
|
| 238b15d54395df53d2219dc12a91e3bfd9c6fa23 |
24-Oct-2015 |
magjed <magjed@webrtc.org> |
SurfaceViewRenderer: Remove use of quitSafely() because it's API lvl 18
I replaced quitSafely() with a CountDownLatch. The reason for not using ThreadUtils.invokeUninterruptibly() is that I want to stop accepting frames asap, and invokeUninterruptibly() would still accept frames during the waiting time.
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1418223002
Cr-Commit-Position: refs/heads/master@{#10393}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| c3402fc3ef0dfebdc82f71ca0bc9a8b549539a2b |
24-Oct-2015 |
magjed <magjed@webrtc.org> |
EGL10.eglCreateWindowSurface(): Replace Surface input with SurfaceHolder
Sending a Surface as input to EGL10.eglCreateWindowSurface() is not supported everywhere. See this code as reference:
https://android.googlesource.com/platform/frameworks/native.git/+/ae9610220b5f509687b840532f95f3638ee0146b/opengl/tools/glgen/stubs/egl/eglCreateWindowSurface.java#42
Sending a SurfaceHolder as input instead should hopefully be supported everywhere, and this is also what GlSurfaceView does:
http://grepcode.com/file/repository.grepcode.com/java/ext/com.google.android/android/5.1.1_r1/android/opengl/GLSurfaceView.java#1076
Review URL: https://codereview.webrtc.org/1416213004
Cr-Commit-Position: refs/heads/master@{#10392}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 49e196af4060624d620297a6bc017699daa33550 |
23-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoFrameType aliases for FrameType.
No longer used in Chromium, so these can now be removed.
BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1415693002 .
Cr-Commit-Position: refs/heads/master@{#10390}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 8c425aa8f66fc2f06df402a0f2163cb53373856f |
23-Oct-2015 |
magjed <magjed@webrtc.org> |
Android: Replace EGL14 with EGL10
The purpose with this change is to support older API levels by replacing EGL14 (API lvl 17) with EGL10 (API lvl 1). The main purpose is to lower API lvl requirement for SurfaceViewRenderer from API lvl 17 to API lvl 15. Also, camera texture capture will work on API lvl < 17 (and texture encode/decode in MediaCodec, but we don't use MediaCodec below API lvl 18?).
GLSurfaceView/VideoRendererGui is already using EGL10.
EGL 1.1 - 1.4 added new functionality, but won't affect performance. We don't need the functionality, so there should be no reason to not use EGL 1.0.
I have profiled AppRTCDemo with Qualcomm Trepn Profiler on a Nexus 5 and Nexus 6 and couldn't see any difference.
Specifically, this CL:
* Update EglBase to use EGL10 instead of EGL14.
* Update imports from EGL14 to EGL10 in a lot of files (plus changing import order in some cases).
* Update VideoCapturerAndroid to always support texture capture.
Review URL: https://codereview.webrtc.org/1396013004
Cr-Commit-Position: refs/heads/master@{#10378}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| ff134ebd3d35ae2edd6eaa63b0a19cb16cc256b7 |
23-Oct-2015 |
tfarina <tfarina@chromium.org> |
talk: Use NDEBUG macro.
NDEBUG is a standard macro with the semantic "Not Debug" for C89, C99, C++98,
C++2003, C++2011, C++2014 standards. There are no _DEBUG macros in the
standards.
_DEBUG is a macro Visual Studio defines when you specify the /MTd or /MDd
option.
http://stackoverflow.com/a/29253284/5237416
This should help fix the TODO in third_party/libjingle/libjingle.gyp
BUG=None
R=sergeyu@chromium.org
Review URL: https://codereview.webrtc.org/1419733004
Cr-Commit-Position: refs/heads/master@{#10377}
pp/webrtc/objc/public/RTCLogging.h
edia/base/videoframe_unittest.h
edia/base/videorenderer.h
ession/media/srtpfilter.cc
|
| c80741f8957b537e968397ac54ff5b5df8a2c318 |
22-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing some issues with the direction attribute of m-lines in offers.
By default, we'll now offer to receive if already receiving
(meaning that the last remote description contained a track).
Also, m-lines that are neither receiving nor sending are now correctly
marked "inactive".
Also moved some logic relating to default tracks out of webrtcsdp.cc,
such that now the direction seen by upper layers will always be
consistent with the consumed/produced SDP.
BUG=528089
Review URL: https://codereview.webrtc.org/1406803004
Cr-Commit-Position: refs/heads/master@{#10376}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/mediasession.cc
|
| 797ef123249f793655640e8cb6ff1eb4fe7e3223 |
22-Oct-2015 |
ivoc <ivoc@webrtc.org> |
Added StopAecDump function to PeerConnectionFactory.
The function to stop recording an AEC dump was missing from the PeerConnectionFactory interface (only a start function was provided). This CL adds the missing stop function.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1415733005
Cr-Commit-Position: refs/heads/master@{#10372}
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 4f4ec0a9270a8cefadfa12e9fa3b979b58b15392 |
22-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 93ea78bfc2c5c661e139f7963d3bc94262c97680 |
22-Oct-2015 |
Henrik Kjellander <kjellander@google.com> |
Add test resources to libjingle_media_unittest.isolate
These should be the final missing pieces before
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux64%20Release%20%28swarming%29
can go green.
BUG=chromium:497757
TBR=stip@chromium.org
Review URL: https://codereview.webrtc.org/1413973004 .
Cr-Commit-Position: refs/heads/master@{#10366}
ibjingle_media_unittest.isolate
|
| 9589e2af1642fce385fb8c47e3726a5c416a4e02 |
22-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Update isolate files for swarming tests
Xvfb is needed for the screen capture tests in modules_unittests,
which also brings in xdisplaycheck used by testing/xvfb.py.
libjingle_media_unittest was missing a resource video in the .isolate
file.
BUG=chromium:497757
R=stip@chromium.org
Review URL: https://codereview.webrtc.org/1415603005 .
Cr-Commit-Position: refs/heads/master@{#10365}
ibjingle_media_unittest.isolate
|
| c96df779b0c9255f25dc78c20a4cd4dff1776384 |
21-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel.
- Remove WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
- Create webrtc::AudioSendStreams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1415563003
Cr-Commit-Position: refs/heads/master@{#10361}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| dfa2815b4f606a58ede5c0214e08a1d5d26d3639 |
21-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Update receive report SSRCs on RemoveSendStream.
Prevents RTCP receiver reports, including PLIs with an old
receiver-report SSRC, from being dropped from the remote sender's
BundleFilter due to no longer being in use.
BUG=chromium:523928, webrtc:4883
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1404363003 .
Cr-Commit-Position: refs/heads/master@{#10359}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| b64a32bf25367d3a32a081680cb9f1972e06759a |
21-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Remove old VideoFrame::Reset.
Hopefully all external implementations are updated, I could build
Chromium locally with this patch. This Reset implementation causes (for
some mysterious reason) -WError=overloaded-virtual failures when trying
to build libjingle APKs.
R=guoweis@webrtc.org, magjed@webrtc.org, pthatcher@webrtc.org
BUG=webrtc:2365
Review URL: https://codereview.webrtc.org/1411253002 .
Cr-Commit-Position: refs/heads/master@{#10352}
edia/base/videoframe.h
|
| 3b7c7935749a955996575e11e718603e4c8cd3a6 |
21-Oct-2015 |
hbos <hbos@webrtc.org> |
New DtlsIdentityStoreInterface::RequestIdentity added that takes rtc::KeyParams. The old RequestIdentity still exists that take rtc::KeyType.
Default implementation added that invokes the other RequestIdentity method, adding default parameters or dropping the parameters.
This CL is in preparation for removing the RequestIdentity that takes rtc::KeyType, necessary as to not break Chromium.
BUG=webrtc:4927, 528250
Review URL: https://codereview.webrtc.org/1414243003
Cr-Commit-Position: refs/heads/master@{#10351}
pp/webrtc/dtlsidentitystore.h
|
| 86b016027d2d27c62fedd108354a2b1274118ae3 |
21-Oct-2015 |
asapersson <asapersson@webrtc.org> |
Add stats for average QP per frame for VP8 (for received video streams):
"WebRTC.Video.Decoded.VP8.Qp"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1340623002
Cr-Commit-Position: refs/heads/master@{#10349}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| e4f96501fc5b3e6de0d1ccd262372afcda1f5b4f |
21-Oct-2015 |
tommi <tommi@webrtc.org> |
Remove system_wrappers/interface/trace_event.h
BUG=
Review URL: https://codereview.webrtc.org/1417773002
Cr-Commit-Position: refs/heads/master@{#10346}
edia/webrtc/webrtcvideoengine2.cc
|
| 0a617e22a46d476abcaaa081cc90300d335da9f9 |
21-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove the default send channel in WVoE.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364643003
Cr-Commit-Position: refs/heads/master@{#10344}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 30a5b5e9fb574016ced1a45ae43921c1a01860a0 |
20-Oct-2015 |
olka <olka@webrtc.org> |
passing |buffer| by reference in AndroidVideoCapturer::OnIncomingFrame
BUG=webrtc:5062
Review URL: https://codereview.webrtc.org/1414703002
Cr-Commit-Position: refs/heads/master@{#10342}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
|
| 43e83d44f01683fbd304e37d47d2f6db0d52660d |
20-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.
Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1411083006
Cr-Commit-Position: refs/heads/master@{#10340}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| a457752f4afc496ed7f4d6b584b08d8635f18cc0 |
20-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10338}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| c6aec4b8edba5c003966a53f20b5e26cb8c7b8ce |
20-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Fix HW video codec stack traces reporting.
Print stack traces for active instance only.
Also add Nexus 4 to H.264 encoder blacklist.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1412833004 .
Cr-Commit-Position: refs/heads/master@{#10329}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| dd2bd26b6d80fe23147ad68dc9ede020b0c4337c |
19-Oct-2015 |
tkchin <tkchin@webrtc.org> |
Update iOS merge script.
BUG=
Review URL: https://codereview.webrtc.org/1414573002
Cr-Commit-Position: refs/heads/master@{#10326}
uild/merge_ios_libs
uild/merge_ios_libs.gyp
|
| 023f3ef0296511f12897c503d6fc2ed063712474 |
19-Oct-2015 |
honghaiz <honghaiz@webrtc.org> |
Create network change notifier and pass the event to NetworkManager
BUG=
Review URL: https://codereview.webrtc.org/1391703003
Cr-Commit-Position: refs/heads/master@{#10325}
pp/webrtc/androidtests/AndroidManifest.xml
pp/webrtc/androidtests/src/org/webrtc/NetworkMonitorTest.java
pp/webrtc/java/android/org/webrtc/NetworkMonitor.java
pp/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java
pp/webrtc/java/jni/androidnetworkmonitor_jni.cc
pp/webrtc/java/jni/androidnetworkmonitor_jni.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/peerconnectioninterface.h
ibjingle.gyp
|
| f56eca031cf64ecaa3a0ae3f77f8a2e14b09092e |
19-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Remove dummyinstantiation.cc.
Prevented Android libjingle APK building since EnsureAPIMatch is defined
but not used.
BUG=webrtc:2365
R=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1410643005 .
Cr-Commit-Position: refs/heads/master@{#10323}
ibjingle_tests.gyp
edia/webrtc/dummyinstantiation.cc
|
| 22993e1a0c114122fc1b9de0fc74d4096ec868bd |
19-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.
BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1371043003
Cr-Commit-Position: refs/heads/master@{#10320}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
edia/webrtc/fakewebrtcvideoengine.h
|
| 9781152e043e35e2f676ddcf5de079c9548f3b37 |
16-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Add new Android camera events.
Add events to track when camera is requested to open,
when first camera frame is available and when camera is
closed.
BUG=b/24271359
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1398793005 .
Cr-Commit-Position: refs/heads/master@{#10306}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| be16f79818d7c21b747189b3e86d8d98add3e6b1 |
16-Oct-2015 |
pbos <pbos@webrtc.org> |
Remove simulcast bitrate modes.
Instead always use the SBM_VERY_HIGH setting.
BUG=webrtc:4885
R=hta@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1407693005
Cr-Commit-Position: refs/heads/master@{#10305}
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/simulcast.cc
edia/webrtc/simulcast.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 12f680214e28dc5f0a13ac8afc0d1445f89e67e6 |
16-Oct-2015 |
perkj <perkj@webrtc.org> |
Revert "Prepare MediaCodecVideoEncoder for surface textures."
This reverts commit 90754174d98d6b71fd4aaed897bd54980f7e59c4.
Revert "Fix use of scaler in MediaCodecVideoEncoder"
This reverts commit ec93628e75fdb81f23635b39b5f3da846bcefd21.
R=magjed@webrtc.org
TBR=glaznev@webrtc.org
BUG=webrtc:4993 b/24984012
Review URL: https://codereview.webrtc.org/1407263002 .
Cr-Commit-Position: refs/heads/master@{#10300}
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 112a3d81db02d349af0ce6c0827da6d8fbc421a8 |
16-Oct-2015 |
ivoc <ivoc@webrtc.org> |
Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1374253002
Cr-Commit-Position: refs/heads/master@{#10297}
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| cbc9507755e730a7f8d81ab3d8cf6efb6678f2ae |
16-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Temporarily rename P2PTestConductor.
Need to do this because some build bots were relying on the previous
name, in order to skip tests that were expected to time out.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1412553002
Cr-Commit-Position: refs/heads/master@{#10295}
pp/webrtc/peerconnection_unittest.cc
|
| 5e97fb5c996743a4c137a5279be6eb6485225b65 |
15-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Don't create remote streams if m-line direction doesn't include "send".
BUG=webrtc:5054
Review URL: https://codereview.webrtc.org/1403173002
Cr-Commit-Position: refs/heads/master@{#10293}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface_unittest.cc
|
| af1b59cf271854177915342692a78ec0aba61ccd |
15-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Cleaning up peerconnection_unittest.
Merging the PeerConnectionTestClientBase and JsepTestClient classes,
since there's no real logical distinction. This should make it slightly
less painful to write new PeerConnection tests.
Review URL: https://codereview.webrtc.org/1393223005
Cr-Commit-Position: refs/heads/master@{#10292}
pp/webrtc/peerconnection_unittest.cc
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
pp/webrtc/fakemediacontroller.h
pp/webrtc/mediacontroller.cc
pp/webrtc/mediacontroller.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/constants.cc
edia/base/constants.h
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/mediachannel.h
edia/base/rtpdataengine.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager_unittest.cc
|
| 543b6ca30a43eeb069c699291460ce6bacc7959d |
15-Oct-2015 |
magjed <magjed@webrtc.org> |
Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
The code that depends on the reverted CL is disabled but not removed. NativeHandleImpl is reverted to the previous implementation, and the new implementation is renamed to NativeTextureHandleImpl. Texture capture can not be used anymore, because it will crash in peerconnection_jni.cc.
Reason for revert:
Increased HW decoder latency and crashes related to that. Also suspected cause of video tearing.
Original issue's description:
> This CL should be the last one in a series to finally
> unblock camera texture capture.
>
> The SurfaceTexture.updateTexImage() calls are moved from
> the video renderers into MediaCodecVideoDecoder, and the
> destructor of the texture frames will signal
> MediaCodecVideoDecoder that the frame has returned. This
> CL also removes the SurfaceTexture from the native handle
> and only exposes the texture matrix instead, because only
> the video source should access the SurfaceTexture.
>
> BUG=webrtc:4993
> R=glaznev@webrtc.org, perkj@webrtc.org
>
> Committed: https://crrev.com/91b348c7029d843e06868ed12b728a809c53176c
> Cr-Commit-Position: refs/heads/master@{#10203}
TBR=glaznev
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1394103005
Cr-Commit-Position: refs/heads/master@{#10288}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| d59daf8023286d63a1b6c8af82eedb684181c1eb |
15-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Merging BaseSession code into WebRtcSession.
After the TransportController CL, BaseSession does little more than
hold a state and an error, and act as an intermediary for the
TransportController. So it doesn't make sense for it to be its own
class.
Review URL: https://codereview.webrtc.org/1397973002
Cr-Commit-Position: refs/heads/master@{#10281}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/channelmanager.h
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor.h
ession/media/currentspeakermonitor_unittest.cc
|
| ab9b2d1516cad017c6e0236c468934582530c965 |
14-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
Reason for reland:
The original CL actually didn't break browser_tests; it was
just a coincidence that it started failing.
Original issue's description:
> Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
>
> Reason for revert:
> Broke browser_tests on Mac. Still need to investigate the cause.
>
> Original issue's description:
> > Moving MediaStreamSignaling logic into PeerConnection.
> >
> > This needs to happen because in the future, m-lines will be offered
> > based on the set of RtpSenders/RtpReceivers, rather than the set of
> > tracks that MediaStreamSignaling knows about.
> >
> > Besides that, MediaStreamSignaling was a "glue class" without
> > a clearly defined role, so it going away is good for other
> > reasons as well.
> >
> > Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> > Cr-Commit-Position: refs/heads/master@{#10268}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/fc648b6d934e936f4d9a32c813364b331536ec3b
> Cr-Commit-Position: refs/heads/master@{#10269}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1404473005
Cr-Commit-Position: refs/heads/master@{#10277}
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ibjingle_tests.gyp
|
| 6d387c0e92f033e31c8dd1efbf3f98bf159c6cf1 |
14-Oct-2015 |
magjed <magjed@webrtc.org> |
Android MediaCodecVideoDecoder: Limit max pending frames to number of input buffers
This CL should reduce the number of timeouts for dequeueInputBuffer() which results in the log "MediaCodecVideo: dequeueInputBuffer error" followed by software fallback for VP8/VP9 and codec restart for H264.
A timeout always happen for dequeueInputBuffer() when frames_received_ > frames_decoded_ + num_input_buffers. The following code tries to drain the decoder before enqueuing more input buffers:
// Try to drain the decoder and wait until output is not too
// much behind the input.
if (frames_received_ > frames_decoded_ + max_pending_frames_) {
ALOGV("Received: %d. Decoded: %d. Wait for output...",
frames_received_, frames_decoded_);
if (!DeliverPendingOutputs(jni, kMediaCodecTimeoutMs,
true /* dropFrames */)) {
ALOGE << "DeliverPendingOutputs error";
return ProcessHWErrorOnCodecThread();
}
if (frames_received_ > frames_decoded_ + max_pending_frames_) {
ALOGE << "Output buffer dequeue timeout";
return ProcessHWErrorOnCodecThread();
}
...
}
However, for H264, |max_pending_frames_| can currently be larger than the number of input buffers so that the code above is never executed. This CL limits |max_pending_frames_| to the number of input buffers.
TBR=glaznev
BUG=b/24867188,b/24864151
Review URL: https://codereview.webrtc.org/1394303005
Cr-Commit-Position: refs/heads/master@{#10273}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
|
| fc648b6d934e936f4d9a32c813364b331536ec3b |
14-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
Reason for revert:
Broke browser_tests on Mac. Still need to investigate the cause.
Original issue's description:
> Moving MediaStreamSignaling logic into PeerConnection.
>
> This needs to happen because in the future, m-lines will be offered
> based on the set of RtpSenders/RtpReceivers, rather than the set of
> tracks that MediaStreamSignaling knows about.
>
> Besides that, MediaStreamSignaling was a "glue class" without
> a clearly defined role, so it going away is good for other
> reasons as well.
>
> Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> Cr-Commit-Position: refs/heads/master@{#10268}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1403633005
Cr-Commit-Position: refs/heads/master@{#10269}
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ibjingle_tests.gyp
|
| 97c392935411398b506861601c82e31d95c591f0 |
13-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Moving MediaStreamSignaling logic into PeerConnection.
This needs to happen because in the future, m-lines will be offered
based on the set of RtpSenders/RtpReceivers, rather than the set of
tracks that MediaStreamSignaling knows about.
Besides that, MediaStreamSignaling was a "glue class" without
a clearly defined role, so it going away is good for other
reasons as well.
Review URL: https://codereview.webrtc.org/1393563002
Cr-Commit-Position: refs/heads/master@{#10268}
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ibjingle_tests.gyp
|
| a0751c5c068ee76aaeeac56173ca043da1d568ff |
13-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Cleanup OWNERS of talk/app/webrtc.
R=juberti@google.com
Review URL: https://codereview.webrtc.org/1404533003 .
Cr-Commit-Position: refs/heads/master@{#10267}
pp/webrtc/OWNERS
|
| 73f44f6481b2a767c80693224ec3b334c26bc4e7 |
13-Oct-2015 |
perkj <perkj@webrtc.org> |
VideoCapturerAndroid, only you SurfaceViewHelper when capturing to textures.
SurfaceViewHelper requires EGL14 that was added in API level 17. Since the SurfaceViewHelper is only neeed when we capture to textures, this cl change back to not use it when we are capturing to byte buffers.
Also, thread.quitsafely was added in level 18. Instead a new ThreadUtil method has been added for this.
BUG=b/24782220
TEST = run
ninja -C out/Debug libjingle_peerconnection_android_unittest && CHECKOUT_SOURCE_ROOT=`pwd` build/android/adb_install_apk.py --debug out/Debug/apks/libjingle_peerconnection_android_unittest.apk && ./third_party/android_tools/sdk/platform-tools/adb shell am instrument -w -e class org.webrtc.VideoCapturerAndroidTest org.webrtc.test/android.test.InstrumentationTestRunner on a device running Android 4.1 (I tried Nexus 7, the first version)
Review URL: https://codereview.webrtc.org/1401023003
Cr-Commit-Position: refs/heads/master@{#10265}
pp/webrtc/androidtests/AndroidManifest.xml
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| ec93628e75fdb81f23635b39b5f3da846bcefd21 |
13-Oct-2015 |
perkj <perkj@webrtc.org> |
Fix use of scaler in MediaCodecVideoEncoder
This bug fixes an issue introduced in https://codereview.webrtc.org/1396073003/
BUG=webrtc:5067
TEST= set new_bit_rate = 200 in MediaCodecVideoEncoder::SetRatesOnCodecThread and compile and run ApprtDemo
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1401943002 .
Cr-Commit-Position: refs/heads/master@{#10263}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 1ac561447e3e1d81a1e390f95a385b5ed8fe0932 |
13-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 3.
Get rid of default receive channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1385893002
Cr-Commit-Position: refs/heads/master@{#10262}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
|
| 8fb30c328b7b5e1ad33e970d1dabca55fdc18926 |
13-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 2.
Rename voe_channel_ to default_send_channel_id_.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1388733002
Cr-Commit-Position: refs/heads/master@{#10261}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 325d414e8cc0990328c49c1d1f4a977f82f90e6a |
12-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Add option to print peer connection factory Java stack traces.
Removing static declaration for media codec thread to allow
running multiple HW codec instances.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1393203005 .
Cr-Commit-Position: refs/heads/master@{#10258}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| a5b62d987a87c16f0a0badbd9b71f9f106ad1cca |
12-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Replace API v23 calls.
R=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1396373002 .
Cr-Commit-Position: refs/heads/master@{#10257}
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
|
| fc950848e39c5627c9af3e3a44e5177843b03d09 |
12-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Fix: RefCountInterface: Make AddRef() and Release() const
The landed CL contained some unwanted changes.
TBR=tommi
Review URL: https://codereview.webrtc.org/1401743002 .
Cr-Commit-Position: refs/heads/master@{#10255}
pp/webrtc/dtmfsender_unittest.cc
|
| 52a30e31f13f6bd28b22f85613f6fd9d25086be2 |
12-Oct-2015 |
magjed <magjed@webrtc.org> |
Reland of Android: Put common native VideoFrameBuffer implementation in androidvideocapturer_jni (patchset #1 id:1 of https://codereview.webrtc.org/1389283003/ )
Reason for revert:
Nothing wrong with the original CL, the bug was in rtc::Bind(), which is fixed now (https://codereview.webrtc.org/1403683004/).
Original issue's description:
> Revert of Android: Put common native VideoFrameBuffer implementation in androidvideocapturer_jni (patchset #1 id:1 of https://codereview.webrtc.org/1391403004/ )
>
> Reason for revert:
> Crashes on AppRTCDemo disconnect
>
> Original issue's description:
> > Android: Put common native VideoFrameBuffer implementation in native_handle_impl.cc
> >
> > BUG=webrtc:4993
> > R=perkj@webrtc.org
> >
> > Committed: https://crrev.com/60472216da0644b49ed5f9fa51c51d4874afafa7
> > Cr-Commit-Position: refs/heads/master@{#10248}
>
> TBR=perkj@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4993
>
> Committed: https://crrev.com/962c26bfd6c3eb3cf7402daaab89404ae38dd534
> Cr-Commit-Position: refs/heads/master@{#10249}
TBR=perkj@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1397373002
Cr-Commit-Position: refs/heads/master@{#10254}
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
|
| 1b40a9a8afe0d7b2244ad8dea19e8222fec3c207 |
12-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
RefCountInterface: Make AddRef() and Release() const
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.
The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:
void Foo::Bar() const {}
rtc::Callback0<void> Foo::MakeClosure() const {
return rtc::Bind(&Foo::Bar, this);
}
We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.
A test is also added in bind_unittest to check this behaviour.
BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1403683004 .
Cr-Commit-Position: refs/heads/master@{#10253}
pp/webrtc/dtmfsender_unittest.cc
edia/webrtc/fakewebrtcvideocapturemodule.h
|
| 90754174d98d6b71fd4aaed897bd54980f7e59c4 |
12-Oct-2015 |
perkj <perkj@webrtc.org> |
Prepare MediaCodecVideoEncoder for surface textures.
This make small refactorings to MediaVideoEncoder to prepare for adding support to encode from textures. The C++ layer does not have any functional changes.
- Moves ResetEncoder to always work on the codec thread
- Adds use of ThreadChecker.
- Change Java MediaEncoder.Init to return true or false and introduce method getInputBuffers.
- Add simple unit test for Java MediaCodecVideoEncoder.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1396073003
Cr-Commit-Position: refs/heads/master@{#10250}
pp/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 962c26bfd6c3eb3cf7402daaab89404ae38dd534 |
12-Oct-2015 |
magjed <magjed@webrtc.org> |
Revert of Android: Put common native VideoFrameBuffer implementation in androidvideocapturer_jni (patchset #1 id:1 of https://codereview.webrtc.org/1391403004/ )
Reason for revert:
Crashes on AppRTCDemo disconnect
Original issue's description:
> Android: Put common native VideoFrameBuffer implementation in native_handle_impl.cc
>
> BUG=webrtc:4993
> R=perkj@webrtc.org
>
> Committed: https://crrev.com/60472216da0644b49ed5f9fa51c51d4874afafa7
> Cr-Commit-Position: refs/heads/master@{#10248}
TBR=perkj@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1389283003
Cr-Commit-Position: refs/heads/master@{#10249}
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
|
| 60472216da0644b49ed5f9fa51c51d4874afafa7 |
12-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Put common native VideoFrameBuffer implementation in native_handle_impl.cc
BUG=webrtc:4993
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1391403004 .
Cr-Commit-Position: refs/heads/master@{#10248}
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
|
| 747c1bccd961a88285d6bfeebcec0cb25f719dfb |
12-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceTextureHelper: Replace API 21 with API 11 version of setOnFrameAvailableListener()
BUG=b/24809429
R=glaznev@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1395543004 .
Cr-Commit-Position: refs/heads/master@{#10247}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
|
| e9e366875992baf60caf5baaec302e2f1de013b2 |
12-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Add helper function to synchronously execute Callables on Handler
TBR=hbos
Review URL: https://codereview.webrtc.org/1398283002 .
Cr-Commit-Position: refs/heads/master@{#10246}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
|
| 69ddaefbb39975fde3e0bb43e233c357f514213c |
10-Oct-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Add option to print peer connection factory Java stack traces."
This reverts commit b68c5995d1ac84866da45a4ecbb180d8c704ad90.
Reason for reverting: It breaks some Android32 bots.
TBR=glaznev@google.com
Review URL: https://codereview.webrtc.org/1399473003 .
Cr-Commit-Position: refs/heads/master@{#10239}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| b68c5995d1ac84866da45a4ecbb180d8c704ad90 |
09-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Add option to print peer connection factory Java stack traces.
Updated version with better handling of media codec release checks.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1397163002 .
Cr-Commit-Position: refs/heads/master@{#10238}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| d4cec0d8fa7913bc9dfa9137e44cca9098e16698 |
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove MediaChannel::SetRemoteRenderer().
This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1398823003
Cr-Commit-Position: refs/heads/master@{#10237}
pp/webrtc/mediastreamprovider.h
pp/webrtc/rtpreceiver.cc
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channel.cc
ession/media/channel.h
|
| 98c68865e715f693390209adb454ab3a5b6de332 |
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.
BUG=webrtc:4690
Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
Cr-Commit-Position: refs/heads/master@{#10226}
Review URL: https://codereview.webrtc.org/1399553003
Cr-Commit-Position: refs/heads/master@{#10235}
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/webrtcsession_unittest.cc
ibjingle.gyp
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| 4bac9c53da9988741d59753c2d789adb94de5e68 |
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change SetOutputScaling to set a single level, not left/right levels.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397773002
Cr-Commit-Position: refs/heads/master@{#10234}
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| 0b67546d8c080f376565a4c1cedd14947fdbaf2b |
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 1.
Rx AGC config bits copied from https://codereview.webrtc.org/1315903004.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1388723002
Cr-Commit-Position: refs/heads/master@{#10233}
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| fd20bb3e80a7d4270a077b404619d00e2faf7f68 |
09-Oct-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Allow to print Java stack traces in Android camera, renderer and media codec."
Reason for revert: It breaks some Android32 bots.
TBR=glaznev@google.com
Review URL: https://codereview.webrtc.org/1397473004 .
Cr-Commit-Position: refs/heads/master@{#10231}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 2b298de1003dfc6e0e76b8782cab6edaac0ad47c |
09-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Reset media codec thread when Encoder/Decoder object is created.
Review URL: https://codereview.webrtc.org/1389373004 .
Cr-Commit-Position: refs/heads/master@{#10230}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| eefbc3bbd7a6962265b028cf259b5028944561d1 |
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ )
Reason for revert:
Breaks Chrome since its build files were not updated prior to file removal.
Original issue's description:
> - Remove AudioTrackRenderer.
> - Remove AddChannel/RemoveChannel from AudioRenderer interface.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
> Cr-Commit-Position: refs/heads/master@{#10226}
TBR=tommi@webrtc.org,solenberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1393343003
Cr-Commit-Position: refs/heads/master@{#10228}
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/webrtcsession_unittest.cc
ibjingle.gyp
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| f0159a742f39fe746053430a04b844ef2ead0770 |
08-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Allow to print Java stack traces in Android camera, renderer and media codec.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1396873002 .
Cr-Commit-Position: refs/heads/master@{#10227}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 1c0bb386b67835feb5934f503dddfe0912bce3ac |
08-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1399553003
Cr-Commit-Position: refs/heads/master@{#10226}
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/webrtcsession_unittest.cc
ibjingle.gyp
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| 69f576010edae80bc83fbf51fa06c3ee611125e8 |
08-Oct-2015 |
lally <lally@webrtc.org> |
Added parsing of either space or colon for sctp-port.
BUG=https://code.google.com/p/webrtc/issues/detail?id=5039
Review URL: https://codereview.webrtc.org/1395523002
Cr-Commit-Position: refs/heads/master@{#10225}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| e76fb369825382cdd3cc136c0bc22a9b10e8caee |
08-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceViewRenderer: Add tests for onMeasure()
BUG=webrtc:4742
R=hbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1379793003 .
Cr-Commit-Position: refs/heads/master@{#10224}
pp/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| bf2004bc3735c7cd12e345b469c3994b72629047 |
08-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceViewRenderer: Only clear image in release() if initialized
This CL is a small bug fix for "Android SurfaceViewRenderer: Allow to re-init after release() has been called" https://codereview.webrtc.org/1389203003/. It is only possible to clear the last image in release() if init() has been called beforehand.
TBR=hbos
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1396573003 .
Cr-Commit-Position: refs/heads/master@{#10223}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| ac30642461c4f94916741106e3ba3f3b7b670a47 |
08-Oct-2015 |
perkj <perkj@webrtc.org> |
Native changes for VideoCapturerAndroid surface texture support
These are the necessary changes in C++ related to the video capturer necessary to capture to a surface texture.
It does not handle scaling / cropping yet though.
BUG=
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1395673003 .
Cr-Commit-Position: refs/heads/master@{#10218}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
|
| 4382d800d225a3d6ce93011ff9a573e4ff613f35 |
08-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceViewRenderer: Allow to re-init after release() has been called
This CL makes a thorough reset of all variables in release() and clears the last rendered image so that the SurfaceViewRenderer object can be reinitialized with init() and work properly. This CL also removes an implicit assumption that init() is called before surfaceCreated() - now they can be called in any order.
BUG=webrtc:4742
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1389203003 .
Cr-Commit-Position: refs/heads/master@{#10217}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| 6ffc3309dee47b31ba118e41acc89a258c754c33 |
08-Oct-2015 |
Henrik Kjellander <kjellander@google.com> |
Remove references to libpeerconnection.
What used to be the libpeerconnection library is now compiled
statically into the Chromium binary, so clean up references it.
BUG=chromium:482123
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1399513002 .
Cr-Commit-Position: refs/heads/master@{#10216}
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
|
| 3d06eca5e3310ea068625a9c95d492a16854f421 |
08-Oct-2015 |
perkj <perkj@webrtc.org> |
Add support to Capture to a texture instead of memory.
This adds support for capturing to a texture in the Java part of VideoCapturerAndroid.
After this cl, the C++ also needs modification.
https://codereview.webrtc.org/1375953002/ contains the idea and have a working version where textures can be rendered in local preview.
BUG=webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1383413002 .
Cr-Commit-Position: refs/heads/master@{#10213}
pp/webrtc/androidtests/AndroidManifest.xml
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
|
| 335204c550e9570d356d0d6264475ac40c7f92f6 |
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
Reason for revert:
Breaks chrome.
Original issue's description:
> provide RSA2048 as per RFC
>
> BUG=webrtc:4972
>
> Committed: https://crrev.com/0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e
> Cr-Commit-Position: refs/heads/master@{#10209}
TBR=hbos@webrtc.org,juberti@google.com,jbauch@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1397703002
Cr-Commit-Position: refs/heads/master@{#10210}
pp/webrtc/webrtcsession_unittest.cc
|
| 0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e |
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
provide RSA2048 as per RFC
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1329493005
Cr-Commit-Position: refs/heads/master@{#10209}
pp/webrtc/webrtcsession_unittest.cc
|
| fddf6e526c8a49e8d67810881c5bed44e2573474 |
08-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Use WebRTC logging in MediaCodec JNI code.
Also enable HW encoder scaling in AppRTCDemo.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1396653002 .
Cr-Commit-Position: refs/heads/master@{#10205}
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 21622a1d19538ca332fe0250ee550d7beeb29802 |
07-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Add option to print peer connection factory Java stack traces.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1395693002 .
Cr-Commit-Position: refs/heads/master@{#10204}
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 91b348c7029d843e06868ed12b728a809c53176c |
07-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android MediaCodecVideoDecoder: Manage lifetime of texture frames
This CL should be the last one in a series to finally unblock camera texture capture.
The SurfaceTexture.updateTexImage() calls are moved from the video renderers into MediaCodecVideoDecoder, and the destructor of the texture frames will signal MediaCodecVideoDecoder that the frame has returned. This CL also removes the SurfaceTexture from the native handle and only exposes the texture matrix instead, because only the video source should access the SurfaceTexture.
BUG=webrtc:4993
R=glaznev@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1378033003 .
Cr-Commit-Position: refs/heads/master@{#10203}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/dtmfsender_unittest.cc
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/objc/avfoundationvideocapturer.h
pp/webrtc/objc/avfoundationvideocapturer.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/remotevideocapturer.cc
pp/webrtc/remotevideocapturer.h
pp/webrtc/remotevideocapturer_unittest.cc
pp/webrtc/rtpreceiver.cc
pp/webrtc/rtpreceiver.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/videosource.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/base/audioframe.h
edia/base/cpuid.cc
edia/base/executablehelpers.h
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/fakevideocapturer.h
edia/base/fakevideorenderer.h
edia/base/mediachannel.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/rtpdump.cc
edia/base/rtpdump.h
edia/base/rtpdump_unittest.cc
edia/base/rtputils.cc
edia/base/rtputils.h
edia/base/rtputils_unittest.cc
edia/base/streamparams.cc
edia/base/streamparams.h
edia/base/streamparams_unittest.cc
edia/base/testutils.cc
edia/base/testutils.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videocommon.cc
edia/base/videocommon.h
edia/base/videoengine_unittest.h
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/base/yuvframegenerator.cc
edia/base/yuvframegenerator.h
edia/devices/carbonvideorenderer.cc
edia/devices/carbonvideorenderer.h
edia/devices/filevideocapturer.cc
edia/devices/filevideocapturer.h
edia/devices/gdivideorenderer.cc
edia/devices/gtkvideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/devices/linuxdevicemanager.cc
edia/devices/mobiledevicemanager.cc
edia/devices/yuvframescapturer.cc
edia/devices/yuvframescapturer.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/simulcast.cc
edia/webrtc/simulcast.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvideoframefactory_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/audiomonitor.cc
ession/media/audiomonitor.h
ession/media/bundlefilter.cc
ession/media/bundlefilter.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor.h
ession/media/currentspeakermonitor_unittest.cc
ession/media/externalhmac.h
ession/media/mediamonitor.cc
ession/media/mediamonitor.h
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/planarfunctions_unittest.cc
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
ession/media/yuvscaler_unittest.cc
|
| d97ec30ce4f22ba2d88314d67ff44458144a5096 |
07-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1386653002
Cr-Commit-Position: refs/heads/master@{#10194}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| a38e31a054c1f3ac9b42931c58814993281744b2 |
06-Oct-2015 |
Henrik Kjellander <kjellander@google.com> |
Update lower-level codereview.settings files.
Every now and then we get CLs to codereview.webrtc.org
that are created from a Chromium checkout by editing
the code in third_party/webrtc or third_party/libjingle.
By editing these lower-level codereview.settings files,
we instead cause a crash during 'git cl upload', but the
contents of the file will also be printed, which can work
as an error message. The alternative would be to entirely
remove the files.
BUG=
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1389963002 .
Cr-Commit-Position: refs/heads/master@{#10191}
odereview.settings
|
| 4139c0f1c53509ea48c936b58a22a66e63e51fda |
06-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Java binding for RtpSender/RtpReceiver.
The Java PeerConnection maintains a cached list of Java RtpSenders
and RtpReceivers so that the same objects are returned every time
getSenders() or getReceivers() is called. They are disposed of when
PeerConnection.dispose() is called, which will also dispose their
referenced MediaStreamTracks.
Review URL: https://codereview.webrtc.org/1368143002
Cr-Commit-Position: refs/heads/master@{#10189}
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/java/src/org/webrtc/RtpReceiver.java
pp/webrtc/java/src/org/webrtc/RtpSender.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
ibjingle.gyp
|
| 0a6c4ca942f3a25c15c7af64a9515d381c34cd9c |
06-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Catching more errors when parsing ICE server URLs.
Every malformed URL should now produce an error message in JS, rather than
silently failing and possibly printing a warning message to the console (and
possibly crashing).
Also added some unit tests, and made "ParseIceServers" public.
BUG=445002
Review URL: https://codereview.webrtc.org/1344143002
Cr-Commit-Position: refs/heads/master@{#10186}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
|
| 09f1350efaa486d84e3b6fede94ced6aa404a85f |
05-Oct-2015 |
Alex Glaznev <glaznev@google.com> |
Add option to reset Android video renderer first frame flag.
This allows to correctly report first frame event in applications which
use same remote video renderer for multiple calls.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1384353002 .
Cr-Commit-Position: refs/heads/master@{#10176}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 6caafbe5b6b777b309a6eb90a02cf54d5106fb9b |
05-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Convert uint16_t to int for WebRTC cipher/crypto suite.
This is a follow up CL on https://codereview.webrtc.org/1337673002
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1377733004 .
Cr-Commit-Position: refs/heads/master@{#10175}
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
|
| 1b33da1298809535348f42ed4d3db23e393e32ee |
05-Oct-2015 |
perkj <perkj@webrtc.org> |
SurfaceTextureHelper fixes
Fixed a problem where eglBase.makecurrent() could be called after the context had been released if SurfaceTextureHelper was first created and immedately disconnected.
Add the possibility to inject a thread to use instead of creating a new.
BUG= webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1384923002 .
Cr-Commit-Position: refs/heads/master@{#10174}
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
|
| 418503275c52dba03c1c9dffec8aabfacd33421e |
05-Oct-2015 |
perkj <perkj@webrtc.org> |
Add ThreadChecker class to ThreadUtils
This class can be used for checking that method calls are made on the correct thread.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1384303002 .
Cr-Commit-Position: refs/heads/master@{#10173}
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
|
| e0bce240652bcf4031ae61985938e968469d3f53 |
05-Oct-2015 |
perkj <perkj@webrtc.org> |
VideoCapturerAndroid: Add custom nativeCreateVideoCapturer()
This CL shouldn't make any functional changes. It adds a new VideoCapturerAndroid.nativeCreateVideoCapturer() instead of always using VideoCapturer.nativeCreateVideoCapturer(). The purpose is to simplify androidvideocapturer_jni and VideoCapturerAndroid.create(). This way, it is possible to use the ctor instead of VideoCapturerAndroid.init() to initialize variables, and they can be made final etc.
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1360173002 .
Cr-Commit-Position: refs/heads/master@{#10171}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
|
| bc0938e8e7db590b6c01f19e57a3418a9cd52523 |
03-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android VideoRendererGui: Make deep copy of incoming texture frames
VideoRendererGui may need to render incoming frames multiple times. We currently call VideoRenderer.renderFrameDone() while we still hold references to the OES texture. This CL makes a deep copy of the OES texture before calling renderFrameDone(). This will truly release the dependency to the incoming frame, so that video textures sources can rely on the renderFrameDone() callback.
This CL is a part of the plan in https://codereview.webrtc.org/1357923002/.
The texture copy doesn't cause any measurable performance difference on a Nexus 5 using VideoRendererGui in a AppRTCDemo loopback call.
BUG=webrtc:4993
TEST=Revert "Enable SurfaceViewRenderer for AppRTCDemo" https://codereview.webrtc.org/1356603004/ and try AppRTCDemo.
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1370113005 .
Cr-Commit-Position: refs/heads/master@{#10157}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 44bf6f5f67ab6b6677e3cd1cceb954c2426f930a |
03-Oct-2015 |
magjed <magjed@webrtc.org> |
Android MediaCodecVideoDecoder: Split DecoderOutputBufferInfo into DecodedByteBuffer and DecodedTextureBuffer
This CL separates the types and code paths for textures vs byte buffers in MediaCodecVideoDecoder.dequeueOutputBuffer() and MediaCodecVideoDecoder::DeliverPendingOutputs(). The purpose is to prepare for lifetime management of textures received from the SurfaceTexture.
This CL is a part of the plan in https://codereview.webrtc.org/1357923002/.
BUG=webrtc:4993
Review URL: https://codereview.webrtc.org/1379383002
Cr-Commit-Position: refs/heads/master@{#10156}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 7e319372abd4403565aa3d27a2d1bfa069969b19 |
02-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android MediaCodecVideoDecoder: Cleanup to prepare for texture liftime management
This CL should not change the behaviour of the decoder. The purpose is to prepare for lifetime management of textures received from the SurfaceTexture. The main change is to only use exceptions for error signaling in MediaCodecVideoDecoder.dequeueOutputBuffer() and MediaCodecVideoDecoder.releaseOutputBuffer(), not both exceptions and error return values.
BUG=webrtc:4993
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1383983003 .
Cr-Commit-Position: refs/heads/master@{#10148}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 6781ea49a8e6b4ef737097ee0e35fd4dbc00cb07 |
02-Oct-2015 |
Magnus Jedvert <magjed@webrtc.org> |
jni/native_handle_impl.h: Move implementation into .cc file
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1383563003 .
Cr-Commit-Position: refs/heads/master@{#10147}
pp/webrtc/java/jni/native_handle_impl.cc
pp/webrtc/java/jni/native_handle_impl.h
ibjingle.gyp
|
| 1d8a506405734d0cef9653704b036ca4f1388960 |
02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.h
|
| da903eaabbb6c6830efcafc3c2ade1d36f511e43 |
02-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1373903003
Cr-Commit-Position: refs/heads/master@{#10143}
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 5aaa9b4fe454195df1def4ebd36301a706fdd8d8 |
02-Oct-2015 |
peah <peah@webrtc.org> |
Removed unused API functions in AudioProcessing and AudioProcessingModule
BUG=
Review URL: https://codereview.webrtc.org/1379123002
Cr-Commit-Position: refs/heads/master@{#10138}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 5629a1dba2af17d16978c2d70eaf15993da975ab |
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1380103002
Cr-Commit-Position: refs/heads/master@{#10137}
ession/media/channel_unittest.cc
|
| 5b14b42e93f17d0ea57f1f8b3e8224082c514946 |
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove unused SignalMediaError and infrastructure.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1362913004
Cr-Commit-Position: refs/heads/master@{#10133}
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| b09b660c53ff2c499d149e05e5c435f5057273fc |
01-Oct-2015 |
magjed <magjed@webrtc.org> |
Remove cricket::VideoFrame::Set/GetElapsedTime()
This CL is a baby step towards consolidating the timestamps in cricket::VideoFrame and webrtc::VideoFrame, so that we can unify the frame classes in the future.
The elapsed time functionality is not really used. If a video sink wants to know the elapsed time since the first frame they can store the first timestamp themselves and calculate the time delta to later frames. This is already done in all video sinks that need the elapsed time. Having redundant timestamps in the frame classes is confusing and error prone.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1324263004
Cr-Commit-Position: refs/heads/master@{#10131}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/objc/avfoundationvideocapturer.mm
pp/webrtc/videotrack_unittest.cc
edia/base/fakevideocapturer.h
edia/base/fakevideorenderer.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/base/videoframefactory.cc
edia/devices/filevideocapturer.cc
edia/devices/filevideocapturer.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvideoframefactory_unittest.cc
|
| dfc8f4ff8731390828884a0a91b99e51f2950275 |
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1378513003
Cr-Commit-Position: refs/heads/master@{#10130}
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
|
| 0ecf1b2f2141c519d31b23a07f5958e2275ef864 |
01-Oct-2015 |
dchakarov.broadsoft <dchakarov.broadsoft@gmail.com> |
Android focus problem on rear camera.
On some devices (confirmed Samsung) the focus mode is not configured correctly by default.
The fix explicitly set the focus mode to FOCUS_MODE_CONTINUOUS_VIDEO if this mode is supported.
BUG=webrtc:4991
Review URL: https://codereview.webrtc.org/1338773002
Cr-Commit-Position: refs/heads/master@{#10128}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 456696a9c1bbd586701dcca3e4b2695e419a10ba |
01-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Change WebRTC SslCipher to be exposed as number only
This is to revert the change of https://codereview.webrtc.org/1380603005/
TBR=pthatcher@webrtc.org
BUG=523033
Review URL: https://codereview.webrtc.org/1375543003 .
Cr-Commit-Position: refs/heads/master@{#10126}
pp/webrtc/fakemetricsobserver.cc
pp/webrtc/fakemetricsobserver.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 27dc29b0df23eed5034f28d4d5f66ea0bb425d6c |
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
Reason for revert:
This broke chromium.fyi bot.
Original issue's description:
> Change WebRTC SslCipher to be exposed as number only.
>
> This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
>
> For SRTP, currently it's still string internally but is reported as IANA number.
>
> This is used by the ongoing CL https://codereview.chromium.org/1335023002.
>
> BUG=523033
>
> Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943
> Cr-Commit-Position: refs/heads/master@{#10124}
TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=523033
Review URL: https://codereview.webrtc.org/1380603005
Cr-Commit-Position: refs/heads/master@{#10125}
pp/webrtc/fakemetricsobserver.cc
pp/webrtc/fakemetricsobserver.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 4fe3c9a77386598db9abd1f0d6983aefee9cc943 |
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Change WebRTC SslCipher to be exposed as number only.
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
For SRTP, currently it's still string internally but is reported as IANA number.
This is used by the ongoing CL https://codereview.chromium.org/1335023002.
BUG=523033
Review URL: https://codereview.webrtc.org/1337673002
Cr-Commit-Position: refs/heads/master@{#10124}
pp/webrtc/fakemetricsobserver.cc
pp/webrtc/fakemetricsobserver.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 2f8a4cad16726e8478afc6fd873918bd01b9ac32 |
01-Oct-2015 |
tkchin <tkchin@webrtc.org> |
Add OWNERS for ObjC dirs.
BUG=
Review URL: https://codereview.webrtc.org/1379923002
Cr-Commit-Position: refs/heads/master@{#10123}
pp/webrtc/objc/OWNERS
pp/webrtc/objctests/OWNERS
|
| 27551c95744be6e888652b3292b4130cc804f59f |
30-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android RendererCommon: Refactor getSamplingMatrix()
This CL refactors RendererCommon.getSamplingMatrix() so it does not have any dependecy to SurfaceTeture. The purpose is to prepare for a change in how texture frames are represented - only the texture matrix will be exposed, not the SurfaceTexture itself. This CL also adds an extra test for RendererCommon.rotateTextureMatrix().
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1375593002 .
Cr-Commit-Position: refs/heads/master@{#10118}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java
pp/webrtc/java/android/org/webrtc/GlRectDrawer.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| bbda54e6fa3108451fbe83a6b55c30a0f443b532 |
30-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android MediaDecoder: Use frame pool to avoid allocations for non-surface decoding
BUG=webrtc:4993
TEST=To test non-surface path, set 'use_surface_ = false' in androidmediadecoder_jni.cc.
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1374153003 .
Cr-Commit-Position: refs/heads/master@{#10116}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
|
| a67696b3cde98163ee38de8313ee9eddb73c662e |
29-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding PeerConnectionInterface::SetConfiguration method. (patchset #1 id:1 of https://codereview.webrtc.org/1361263002/ )
Reason for revert:
Relanding with SetConfiguration not pure virtual.
Original issue's description:
> Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
>
> Reason for revert:
> Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.
>
> Original issue's description:
> > Adding PeerConnectionInterface::SetConfiguration method.
> >
> > Also updated the JNI and Objective-C bindings. Later, will have a CL to
> > remove UpdateIce, which this method effectively replaces.
> >
> > BUG=webrtc:4945
> >
> > Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> > Cr-Commit-Position: refs/heads/master@{#10040}
>
> TBR=guoweis@webrtc.org,pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4945
>
> Committed: https://crrev.com/7603c76ab077b1e2033bb179595129bd96797345
> Cr-Commit-Position: refs/heads/master@{#10041}
TBR=guoweis@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945
Review URL: https://codereview.webrtc.org/1361273002
Cr-Commit-Position: refs/heads/master@{#10112}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
|
| 24b52f8322ae225f5e95b9f4d33c976add803a81 |
29-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android GlRectDrawer: Add test for OES texture rendering
BUG=webrtc:4742
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1367913003 .
Cr-Commit-Position: refs/heads/master@{#10109}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
|
| 1d640e53bde26f6951e274a0837aeedb82f5ea3f |
29-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
JavaVideoRendererWrapper: Use jlongFromPointer() to convert frame pointer to jlong
The purpose of this CL is to use jlongFromPointer() for converting frame pointers to jlong instead of implicit casts which is not safe.
In order to respect constness, I had to make a small helper function for this.
BUG=webrtc:4993
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1373233002 .
Cr-Commit-Position: refs/heads/master@{#10108}
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 63b345441a995665c1cdd0329b65f895675874ff |
29-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Simplify handling of options in WebRtcVoiceMediaEngine.
Also removes unnecessary typedef ChannelList.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364753002
Cr-Commit-Position: refs/heads/master@{#10107}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| b5815c801362c98d99668924a8d3ba763ec36a04 |
29-Sep-2015 |
magjed <magjed@webrtc.org> |
Revert of Android VideoCapturer: Send ByteBuffer instead of byte[] (patchset #1 id:1 of https://codereview.webrtc.org/1372813002/ )
Reason for revert:
The top row in the video stream from the camera is messed up. The byte[] pointer is not the same as GetDirectBufferAddress() apparently.
Original issue's description:
> Android VideoCapturer: Send ByteBuffer instead of byte[]
>
> The purpose with this CL is to replace GetByteArrayElements() and ReleaseByteArrayElements() with GetDirectBufferAddress().
>
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/cb3649b40b3fd6d5bbb0a92003b717e46ce90924
> Cr-Commit-Position: refs/heads/master@{#10091}
TBR=hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1377783002
Cr-Commit-Position: refs/heads/master@{#10103}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
|
| 70ab1a1ca89d280a7d51e3fadc51d4be9df209ca |
29-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Exposing RtpSenders and RtpReceivers from PeerConnection.
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.
It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.
Review URL: https://codereview.webrtc.org/1351803002
Cr-Commit-Position: refs/heads/master@{#10100}
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
pp/webrtc/rtpreceiver.cc
pp/webrtc/rtpreceiver.h
pp/webrtc/rtpreceiverinterface.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
pp/webrtc/rtpsenderreceiver_unittest.cc
pp/webrtc/test/fakemediastreamsignaling.h
ibjingle.gyp
ibjingle_tests.gyp
|
| 8e9cb09506b4a076bc097324e8b79a72d3124615 |
28-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Add unittests for SurfaceTextureHelper
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1371643002 .
Cr-Commit-Position: refs/heads/master@{#10099}
pp/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
|
| 4fa648be681ee20d43f79f6c9ec9570631ebcf5a |
28-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding 20-second timeout to Java and Objective-C tests.
This is the same sort of thing we do in C++ end-to-end PeerConnection
tests.
Review URL: https://codereview.webrtc.org/1361213002
Cr-Commit-Position: refs/heads/master@{#10098}
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
|
| 2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1369263002
Cr-Commit-Position: refs/heads/master@{#10096}
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.h
|
| 1f429e34180ca19a7fb98b89bacd34d42e9b01ec |
28-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Passing the new policy from PeerConnection RTCConfiguration to
p2ptransportchannel. This CL does not use the new policy yet.
BUG=
Review URL: https://codereview.webrtc.org/1369773003
Cr-Commit-Position: refs/heads/master@{#10092}
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
|
| cb3649b40b3fd6d5bbb0a92003b717e46ce90924 |
28-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android VideoCapturer: Send ByteBuffer instead of byte[]
The purpose with this CL is to replace GetByteArrayElements() and ReleaseByteArrayElements() with GetDirectBufferAddress().
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1372813002 .
Cr-Commit-Position: refs/heads/master@{#10091}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
|
| d2413e514ad9970f9a04e597275291ab95ad8a5c |
28-Sep-2015 |
Per <perkj@chromium.org> |
Fix the C++ SurfaceTextureHolder
This cl moves back loading java SurfaceTextureHolder to the ClassReferenceHolder and use FindClass through ClassReferenceHolder. Without this, jni->FindClass returns nullptr in surfacetexturehelper_jni.cc.
BUG=webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1370013002 .
Cr-Commit-Position: refs/heads/master@{#10086}
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
|
| 1ab271c1c49191e04e3ac9516e6d92cde739a954 |
28-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceTextureHelper: Don't wait for pending frames in disconnect()
This CL also makes some small non-functional changes in ThreadUtils and EglBase to support SurfaceTextures and SurfaceTextureHelper.
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1368093003 .
Cr-Commit-Position: refs/heads/master@{#10085}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
|
| 3e9eb4ba01acf8a49fa1949305fb30a7daf51964 |
28-Sep-2015 |
Per <perkj@chromium.org> |
Add C++ SurfaceTextureHandler
This cl adds a C++ counterpart of the Java SurfaceTextureHandler. It can be used for creating a webrtc::VideoFrames from a native handle and also guarantee that the Java SurfaceTexture is notified when the VideoFrame is no longer in use.
BUG=webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1366413003 .
Cr-Commit-Position: refs/heads/master@{#10084}
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
pp/webrtc/java/jni/surfacetexturehelper_jni.cc
pp/webrtc/java/jni/surfacetexturehelper_jni.h
ibjingle.gyp
|
| d6d27e7340bca1598973e2197cf08d79ce9aeb04 |
25-Sep-2015 |
Henrik Kjellander <kjellander@google.com> |
Update isolate.gypi to support Swarming + move .isolate files
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.
In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.
BUG=497757
R=maruel@chromium.org
TBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1373513002 .
Cr-Commit-Position: refs/heads/master@{#10081}
uild/isolate.gypi
|
| 17417707428816ffb88d9c71dcc8a5d492cf9fdf |
25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Implement a high-QP threshold for Android H.264.
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.
BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1364253002 .
Cr-Commit-Position: refs/heads/master@{#10078}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 88799d9c1f496ecbe2b17c7508c95566175a29fc |
25-Sep-2015 |
christoffer <christoffer@sinch.com> |
RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error.
Fix an issue where using setNeedsDisplay on a GLKView which has a frame
with size zero will make GLKView/iOS output the following error:
Failed to bind EAGLDrawable: <CAEAGLLayer: 0x1742282e0> to
GL_RENDERBUFFER 1 Failed to make complete framebuffer object 8cd6
(The error code 8cd6 corresponds to
GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT.)
GLKView will internally setup it's render buffer when the delegate is
about to draw into it. Previously when enableSetNeedsDisplay was set to
YES (default), then GLKView would still attempt to setup it's internal
buffer even if it's frame size is zero and that would cause
GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT.
By using enableSetNeedsDisplay = NO, RTCEAGLVideoView can guard against
calling -[GLKView display] if it's current frame size is empty.
Review URL: https://codereview.webrtc.org/1347013002
Cr-Commit-Position: refs/heads/master@{#10076}
pp/webrtc/objc/RTCEAGLVideoView.m
|
| 495d2fdd6569c252aebf0bae5c7cab0a207dd6dc |
25-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Reland of "Android GlRectDrawer: Add test for RGB rendering"
Reland of https://codereview.webrtc.org/1367923002/.
The bug was that not all platforms support glReadPixels() with GL_RGB. This CL uses GL_RGBA instead.
BUG=webrtc:4742
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1370653002 .
Cr-Commit-Position: refs/heads/master@{#10070}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
|
| 6979b024d7cebfdcd1e8f66da59ea157bbc9e47e |
25-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding stub files for RtpSender/RtpReceiver.
This will allow Chromium's build files to be updated, so that when the
real RtpSender CL is submitted, it doesn't break the FYI bots.
Review URL: https://codereview.webrtc.org/1364813004
Cr-Commit-Position: refs/heads/master@{#10065}
pp/webrtc/rtpreceiver.cc
pp/webrtc/rtpreceiver.h
pp/webrtc/rtpreceiverinterface.h
pp/webrtc/rtpsender.cc
pp/webrtc/rtpsender.h
pp/webrtc/rtpsenderinterface.h
|
| ea70d77fd54fa9bd0401220bfbca585c575f7eff |
24-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoCapturerAndroid: Add test for making calls on stopped camera
BUG=webrtc:4978
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1350663002 .
Cr-Commit-Position: refs/heads/master@{#10062}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
|
| 34fbfff068bf46d27812fb8fd531aea889a5feaf |
24-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoMediaChannel::SetRender().
Was a no-op in current implementation.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1334793003 .
Cr-Commit-Position: refs/heads/master@{#10059}
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
ession/media/channel.cc
ession/media/channel_unittest.cc
|
| 5e9a1bc79066eb8d8be62a12da921ee7b430ab9f |
24-Sep-2015 |
magjed <magjed@webrtc.org> |
Revert of Android GlRectDrawer: Add test for RGB rendering (patchset #3 id:40001 of https://codereview.webrtc.org/1367923002/ )
Reason for revert:
The test fails on Nexus 9.
Original issue's description:
> Android GlRectDrawer: Add test for RGB rendering
>
> BUG=webrtc:4742
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/6b20ad99e04f594a9a131bea5d80940698e6e8fd
> Cr-Commit-Position: refs/heads/master@{#10050}
TBR=hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1363613003
Cr-Commit-Position: refs/heads/master@{#10058}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
|
| 2bc68c731dd1e643f89d636de4cdb1dc5d63bd40 |
24-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Wire up QualityScaler for H.264 on Android.
BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1365063002 .
Cr-Commit-Position: refs/heads/master@{#10055}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 6b20ad99e04f594a9a131bea5d80940698e6e8fd |
24-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android GlRectDrawer: Add test for RGB rendering
BUG=webrtc:4742
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1367923002 .
Cr-Commit-Position: refs/heads/master@{#10050}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
|
| 2efe58b1892e6f169cdd3ed8426959868ef5497d |
24-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoCapturerAndroidTest: Dispose PeerConnectionFactory with pending frames
Partial revert of change in testReturnBufferLateEndToEnd from https://codereview.webrtc.org/1350863002/. It is ok to dispose PeerConnectionFactory with pending frames after all.
BUG=webrtc:4909
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1363303002 .
Cr-Commit-Position: refs/heads/master@{#10049}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
|
| 4a3ccad29e4f14c4a66d10edda0d364ea415e309 |
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove SetAudioDelayOffset() and friends.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364093002
Cr-Commit-Position: refs/heads/master@{#10047}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 61e933eac7673feb2f8663c3e71e503b714b350f |
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove ChannelManager::GetCapabilities()
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364083002
Cr-Commit-Position: refs/heads/master@{#10045}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| facbbecb516547adc2ac684c8e0be95ad79dfd88 |
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove use of DeviceManager from ChannelManager.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1346153002
Cr-Commit-Position: refs/heads/master@{#10042}
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/videosource_unittest.cc
pp/webrtc/videotrack_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 7603c76ab077b1e2033bb179595129bd96797345 |
24-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
Reason for revert:
Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.
Original issue's description:
> Adding PeerConnectionInterface::SetConfiguration method.
>
> Also updated the JNI and Objective-C bindings. Later, will have a CL to
> remove UpdateIce, which this method effectively replaces.
>
> BUG=webrtc:4945
>
> Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> Cr-Commit-Position: refs/heads/master@{#10040}
TBR=guoweis@webrtc.org,pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945
Review URL: https://codereview.webrtc.org/1361263002
Cr-Commit-Position: refs/heads/master@{#10041}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
|
| 70702afbcb8418fe93747e7ed63bcbf5e56b90e9 |
24-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding PeerConnectionInterface::SetConfiguration method.
Also updated the JNI and Objective-C bindings. Later, will have a CL to
remove UpdateIce, which this method effectively replaces.
BUG=webrtc:4945
Review URL: https://codereview.webrtc.org/1317353005
Cr-Commit-Position: refs/heads/master@{#10040}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
|
| cbecd358e032021eac11fb13e04ec7f070d4f407 |
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/
Which fixes the FYI bots for the original CL, and breaks them for this revert.
Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}
TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1361773005
Cr-Commit-Position: refs/heads/master@{#10036}
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/peerconnection.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| d0b5b091e4f2724ed2aaf79a77d00487e041f642 |
23-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Add myself as OWNER of webrtc/voice_engine and talk/media/webrtc.
BUG=webrtc:4690
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1359983003 .
Cr-Commit-Position: refs/heads/master@{#10035}
edia/webrtc/OWNERS
|
| c14f5ff60fb0c42c97702de112a9e8f1eccba574 |
23-Sep-2015 |
henrika <henrika@webrtc.org> |
Improving support for Android Audio Effects in WebRTC.
Now also supports AGC and NS effects and adds the possibility
to override default settings.
R=magjed@webrtc.org, pbos@webrtc.org, sophiechang@chromium.org
TBR=perkj
BUG=NONE
Review URL: https://codereview.webrtc.org/1344563002 .
Cr-Commit-Position: refs/heads/master@{#10030}
pp/webrtc/test/fakeaudiocapturemodule.h
ibjingle.gyp
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| d5c75b1a0ba1548d3561109e3e5e63757509e9ae |
23-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Reduce LS_INFO spam from voice_engine/.
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).
BUG=
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1347353004 .
Cr-Commit-Position: refs/heads/master@{#10028}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| 7d173362d01229fe262df37e34ecb061aee8edc3 |
23-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove the [Un]RegisterVoiceProcessor() API.
BUG=webrtc:4690
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1361633002 .
Cr-Commit-Position: refs/heads/master@{#10027}
ibjingle.gyp
ibjingle_tests.gyp
edia/base/capturemanager_unittest.cc
edia/base/fakemediaengine.h
edia/base/fakemediaprocessor.h
edia/base/mediaengine.h
edia/base/videocapturer_unittest.cc
edia/base/voiceprocessor.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 09677342ae9dce4f4ec9c294342a8b1789dcdba2 |
23-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.
BUG=webrtc:4690
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1360773002 .
Cr-Commit-Position: refs/heads/master@{#10026}
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| f706c8ae914da976f16205ff13d15b1a28ead8fd |
23-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoCapturerAndroid: Fix threading issues
This CL makes the following changes:
* Instead of creating a new thread per startCapture()/stopCapture() cycle, VideoCapturerAndroid has a single thread that is initialized in the constructor and kept during the lifetime of the instance. This is more convenient because then it is always possible to post runnables without if-checks. This way, a lot of synchronize statements can be removed. Also, when the camera thread is preserved after stopCapture() it is possible to post late returnBuffer() calls to the correct thread.
* FramePool now enforces single thread use and returnBuffer() calls are posted to the camera thread. This is important because the camera should only be used from one thread, and we call camera.addCallbackBuffer() in returnBuffer().
* switchCamera() no longer returns false on failure, but instead signals the result via the callback.
* Update the test testCaptureAndAsyncRender() - it's not a valid use case to have outstanding frames when calling PeerConnectionFactory.dispose(). Instead, the renderer implementations should have release() functions that block until all frames are returned. The release() functions need to be called in the correct order with PeerConnectionFactory.dispose() last.
BUG=webrtc:4909
R=hbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1350863002 .
Cr-Commit-Position: refs/heads/master@{#10025}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
|
| a81a42f584baa0d93a4b93da9632415e8922450c |
23-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1358413003
Cr-Commit-Position: refs/heads/master@{#10024}
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/peerconnection.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 47ee2f3b9f33e8938948c482c921d4e13a3acd83 |
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
Review URL: https://codereview.webrtc.org/1350523003
Cr-Commit-Position: refs/heads/master@{#10022}
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/peerconnection.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| c1a1b353ec96a92f8b88dba5a058af8744e81560 |
22-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove the SetLocalMonitor() API.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1344083004
Cr-Commit-Position: refs/heads/master@{#10020}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 07d09364b003e6738a02d9940aebab5d3814da6d |
22-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Purge nss files and dependencies.
This replaces https://codereview.webrtc.org/1313233005
which was reverted after triggering Chromium issues.
The only difference is that we're cleaned up dependencies
on use_openssl from the gyp file.
Since https://codereview.chromium.org/1358913003 landed,
this CL should cause no Chromium issues.
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1351503004
Cr-Commit-Position: refs/heads/master@{#10019}
edia/sctp/sctpdataengine_unittest.cc
|
| 82122650536e0a96b5d999b635e04a499e5d9b46 |
22-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Add class ThreadUtils with helper function joinUninterruptibly()
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1356293002 .
Cr-Commit-Position: refs/heads/master@{#10016}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/ThreadUtils.java
ibjingle.gyp
|
| 22011c1b54021ec9a2b4885519e5ce995b1300a2 |
22-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle).
BUG=webrtc:4690
TBR=juberti
Review URL: https://codereview.webrtc.org/1325023005
Cr-Commit-Position: refs/heads/master@{#10011}
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| ef5d5e45cb2b13b17f98747ea9b8e8af7b293977 |
22-Sep-2015 |
asapersson <asapersson@webrtc.org> |
Add field trial for automic resize in MediaCodecVideoEncoder.
BUG=webrtc:4968
Review URL: https://codereview.webrtc.org/1351573002
Cr-Commit-Position: refs/heads/master@{#10009}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 04ac81f2fd8ef6680522438fac1894db5415a0ec |
21-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
BUG=4937
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345913004 .
Cr-Commit-Position: refs/heads/master@{#10004}
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 275a2f16fd99b0f1eb43fd4ba62541af14e797c0 |
21-Sep-2015 |
tommi <tommi@webrtc.org> |
Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ )
Reason for revert:
Broke the Windows build:
[226/365] LINK_EMBED cc_perftests.exe
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj.rsp /c ..\..\remoting\protocol\channel_socket_adapter_unittest.cc /Foobj\remoting\protocol\remoting_unittests.channel_socket_adapter_unittest.obj /Fdobj\remoting\remoting_unittests.cc.pdb
e:\b\build\slave\win\build\src\remoting\protocol\channel_socket_adapter_unittest.cc(36) : error C3861: 'set_readable': identifier not found
ninja: build stopped: subcommand failed.
Original issue's description:
> Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
> If a connection does not receive for 30 seconds, it will be deleted.
> BUG=
>
> Committed: https://crrev.com/ae16f8547d3b447f62f6660f13688585c6c3de15
> Cr-Commit-Position: refs/heads/master@{#10001}
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1356103002
Cr-Commit-Position: refs/heads/master@{#10002}
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| ae16f8547d3b447f62f6660f13688585c6c3de15 |
21-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet).
If a connection does not receive for 30 seconds, it will be deleted.
BUG=
Review URL: https://codereview.webrtc.org/1351673003
Cr-Commit-Position: refs/heads/master@{#10001}
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| c19922c181f0df0b5f4a5c7ffe55f6a5eb2e6ced |
21-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android SurfaceViewRenderer: Block in release() until frames are returned and cleanup is done
BUG=webrtc:4742,webrtc:4909
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1354393002 .
Cr-Commit-Position: refs/heads/master@{#10000}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
|
| e6d3adab087281c24076a534b1c82e47e60153bd |
21-Sep-2015 |
Per <perkj@chromium.org> |
Re-add SurfaceTexture as member for setLocalPreview in VideoCapturerAndroid.
The Android camera api requires a surface to be set in order work. In https://codereview.webrtc.org/1354683004/ this surfaceTexture was removed as a member but it turns out that can lead to camera freezes when the device is rotated. This cl re-adds the surface as a member.
BUG= webrtc:5021, webrtc:5003
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1349433003 .
Cr-Commit-Position: refs/heads/master@{#9999}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 780be751902e38687e578b4429995d783dcbae76 |
21-Sep-2015 |
perkj <perkj@webrtc.org> |
Make PeerConnectionTest.doTest wait for ice candidates
This change the PeerConnectionTest.doTest wait for at least one ice candidate and also make sure the list of candidates in gotIceCandidates is synchronized.
BUG=webrtc:5010
Review URL: https://codereview.webrtc.org/1354913002
Cr-Commit-Position: refs/heads/master@{#9997}
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
|
| 35d1767cc3ae1fd48e8fd01b0b8ed9061734538e |
21-Sep-2015 |
perkj <perkj@webrtc.org> |
Remove the video capture module on Android.
Video capture for android is now implemented in talk/app/webrtc/androidvideocapturer.h
BUG=webrtc:4475
Review URL: https://codereview.webrtc.org/1347083003
Cr-Commit-Position: refs/heads/master@{#9995}
ibjingle.gyp
|
| 8902433a43bbc9cc0de4966774d3dbbe37ef96fb |
18-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.
Cr-Commit-Position: refs/heads/master@{#9994}
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/peerconnection.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 9af63f473e1d0d6c47a741a046c41642dfc1c178 |
18-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".
Review URL: https://codereview.webrtc.org/1246913005
Cr-Commit-Position: refs/heads/master@{#9993}
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/peerconnection.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 4a783086b69acc6e31c5244ba983a25c185fa512 |
18-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Add helper class GlTextureFrameBuffer
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1348513003 .
Cr-Commit-Position: refs/heads/master@{#9991}
pp/webrtc/java/android/org/webrtc/GlTextureFrameBuffer.java
ibjingle.gyp
|
| 3520f9e0493b106d71cae3ee67108e00af13f1f7 |
18-Sep-2015 |
Per <perkj@chromium.org> |
Removes camera.setPreviewTexture in doStopCaptureOnCameraThread and removes the try catch statement since the only method throwing an exception was setPreviewTexture.
BUG=webrtc:5003
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1354683004 .
Cr-Commit-Position: refs/heads/master@{#9985}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| 586b19bdb615dde34cdcf107272d8857fe2f5631 |
18-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
Enable probing with repeated payload packets by default.
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.
In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.
BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1327933003 .
Cr-Commit-Position: refs/heads/master@{#9984}
edia/base/fakenetworkinterface.h
edia/base/videoengine_unittest.h
|
| 71df77bba01c86bc3bd359a61573698ea064f35f |
18-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Remove overridden basictypes.h.
* Use (u)intxx_t for (u)intxx typedefs for all platforms.
* Always include stdint.h.
* Add RTC_ prefix to ARCH_XXX macros.
Chromium did the (u)intxx_t change in
https://codereview.chromium.org/117323010 and
https://codereview.chromium.org/639293007
BUG=chromium:468375
TBR=perkj@webrtc.org (for trivial talk/* changes)
NOTRY=true
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1349213003
Cr-Commit-Position: refs/heads/master@{#9983}
edia/base/cpuid.h
edia/base/yuvframegenerator.h
|
| 7cbd188c5ed7df80bb737bd4ada94422730e2d89 |
18-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (again).
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1353713002 .
Cr-Commit-Position: refs/heads/master@{#9979}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| ac547a653862744d0aae560713f8418ad2852085 |
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
edia/webrtc/webrtcvoiceengine.h
|
| e1c5ec72c6a55b12a00ed3de1504c663367c058d |
17-Sep-2015 |
Patrik Höglund <phoglund@webrtc.org> |
Fixing bad merge (CHECK is now RTC_CHECK)
BUG=None
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1346013006 .
Cr-Commit-Position: refs/heads/master@{#9975}
pp/webrtc/java/jni/peerconnection_jni.cc
|
| fdd1b9a58e4ffbd1442f7d5b1a0bc9c602a8ed5f |
17-Sep-2015 |
Patrik Höglund <phoglund@webrtc.org> |
Reland: Bailing out if pc factory fails to get created.
This was reverted, but it turned out GOMA was down.
This prevents us from continuing if we fail initialization.
The failure will happen closer to its source, rather than
when we try to create the first peer connection.
BUG=None
R=glaznev@webrtc.org
Committed: https://crrev.com/6eb75d9e67f02c256436eb96f3c77026486561a1
Cr-Commit-Position: refs/heads/master@{#9948}
Review URL: https://codereview.webrtc.org/1339923004 .
Cr-Commit-Position: refs/heads/master@{#9974}
pp/webrtc/java/jni/peerconnection_jni.cc
|
| b071a19019a0a2173cc139c960d6ef6946a1c581 |
17-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1327933002 .
Cr-Commit-Position: refs/heads/master@{#9973}
pp/webrtc/statscollector_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel_unittest.cc
|
| f2bfc2b8ef3d774658b9ce3dcd6757f932d071fb |
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove some dead code.
WebRtcPassthroughRender has been dead since webrtcvideoengine.cc was
removed, FakeExternalTransport has probably been unused for a long time.
BUG=webrtc:1695
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1343393003 .
Cr-Commit-Position: refs/heads/master@{#9968}
ibjingle.gyp
ibjingle_tests.gyp
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcpassthroughrender.h
edia/webrtc/webrtcpassthroughrender_unittest.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/fakemetricsobserver.cc
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/jni_helpers.cc
pp/webrtc/java/jni/jni_helpers.h
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/mediacontroller.cc
pp/webrtc/objc/RTCFileLogger.mm
pp/webrtc/objc/avfoundationvideocapturer.mm
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/test/fakedtlsidentitystore.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
edia/base/capturemanager.cc
edia/sctp/sctpdataengine.cc
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager_unittest.cc
|
| 7754285f7c2651c564a48de978c41b141ecfcb02 |
17-Sep-2015 |
Jiayang Liu <jiayl@chromium.org> |
Log to the webrtc log stream from webrtc/modules java code.
The purpose is to gather all webrtc logging in a single place and allow the app to redirect all webrtc logging to a single stream for offline debugging.
Moved Logging.java to webrtc/base to be shared by talk/ and modules/.
R=glaznev@webrtc.org, henrika@webrtc.org, magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1335103004 .
Cr-Commit-Position: refs/heads/master@{#9959}
pp/webrtc/java/src/org/webrtc/Logging.java
ibjingle.gyp
|
| 5975b3c5be44de32ee28c34ca577474e36259e6e |
16-Sep-2015 |
Jiayang Liu <jiayl@chromium.org> |
Log to webrtc logging stream from java code.
Future log messages should all be sent to org.webrtc.Logging as well.
BUG=
Committed: https://crrev.com/66f0da2197974dcc1008f25df2bb4e1d463ad506
Cr-Commit-Position: refs/heads/master@{#9936}
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1338033003 .
Cr-Commit-Position: refs/heads/master@{#9957}
pp/webrtc/java/android/org/webrtc/Camera2Enumerator.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/CameraEnumerator.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/GlShader.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 384194369b4be41912353631a68689129a49e58c |
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
pp/webrtc/videosource.cc
edia/base/streamparams.h
|
| 3c089d751ede283e21e186885eaf705c3257ccd2 |
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
pp/webrtc/dtmfsender.h
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription.h
pp/webrtc/mediacontroller.cc
pp/webrtc/remotevideocapturer.h
pp/webrtc/statstypes.h
pp/webrtc/videosource.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/base/cpuid.h
edia/base/rtpdump.h
edia/base/streamparams.h
edia/base/videoadapter.h
edia/base/videocapturer.h
edia/base/yuvframegenerator.h
edia/devices/filevideocapturer.cc
edia/devices/filevideocapturer.h
edia/devices/yuvframescapturer.cc
edia/devices/yuvframescapturer.h
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 207370f0a2853cc79fb0e0a18cf96ae5c1748c28 |
16-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android MediaCodecVideoDecoder: Remove redundant useSurface arguments
This CL should not do any functional changes. It removes some redundant arguments and unnecessary error checking.
BUG=webrtc:4993
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1338943003 .
Cr-Commit-Position: refs/heads/master@{#9950}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 01ddf01d9c787601064a28a01ce60476aeeaa9d0 |
16-Sep-2015 |
phoglund <phoglund@webrtc.org> |
Revert of Bailing out if pc factory fails to get created. (patchset #1 id:1 of https://codereview.webrtc.org/1339923004/ )
Reason for revert:
Breaks goma (??!??!?)
Original issue's description:
> Bailing out if pc factory fails to get created.
>
> This prevents us from continuing if we fail initialization.
> The failure will happen closer to its source, rather than
> when we try to create the first peer connection.
>
> BUG=None
> R=glaznev@webrtc.org
>
> Committed: https://crrev.com/6eb75d9e67f02c256436eb96f3c77026486561a1
> Cr-Commit-Position: refs/heads/master@{#9948}
TBR=glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review URL: https://codereview.webrtc.org/1344363002
Cr-Commit-Position: refs/heads/master@{#9949}
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 6eb75d9e67f02c256436eb96f3c77026486561a1 |
16-Sep-2015 |
Patrik Höglund <phoglund@webrtc.org> |
Bailing out if pc factory fails to get created.
This prevents us from continuing if we fail initialization.
The failure will happen closer to its source, rather than
when we try to create the first peer connection.
BUG=None
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1339923004 .
Cr-Commit-Position: refs/heads/master@{#9948}
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 2338fec627dd5fa8e0cfe6ebccee2cb87fc4bf38 |
16-Sep-2015 |
Alex Glaznev <glaznev@google.com> |
Partial revert of r9936.
Need to figure out the best way to initialize native logging system
while peer connection factory is not created yet.
R=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1343163003 .
Cr-Commit-Position: refs/heads/master@{#9947}
pp/webrtc/java/android/org/webrtc/Camera2Enumerator.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/CameraEnumerator.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/GlShader.java
pp/webrtc/java/android/org/webrtc/GlUtil.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 32b5d231775c7862da74bd817ab00a54bb85a9b8 |
15-Sep-2015 |
Alex Glaznev <glaznev@google.com> |
Add an option to avoid Java video track release when peer connection is closed.
Currently disposing Java peer connection object will result in auto
release of media streams and media tracks added to peer connection.
Add an option to allow application to own video track so it can be
kept after peer connection is destroyed.
R=jiayl@webrtc.org, wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1333363002 .
Cr-Commit-Position: refs/heads/master@{#9946}
pp/webrtc/java/src/org/webrtc/MediaStream.java
|
| 709ed67c38d0a942f3bf3e68e337cc27a27bc353 |
15-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1269863005 .
Cr-Commit-Position: refs/heads/master@{#9939}
pp/webrtc/mediacontroller.cc
pp/webrtc/mediacontroller.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ibjingle.gyp
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 4ae28a10701c82e7bcdd32f075bb89f43d745c95 |
15-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Add SurfaceTextureHelper for creating and managing SurfaceTextures
Add new helper class to create and synchronize access to SurfaceTextures. The plan is replace the SurfaceTexture in MediaCodecVideoDecoder in a follow-up CL and remove the SurfaceTexture.updateTexImage() call in VideoRendererGui.
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1342713003 .
Cr-Commit-Position: refs/heads/master@{#9938}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
ibjingle.gyp
|
| 66f0da2197974dcc1008f25df2bb4e1d463ad506 |
15-Sep-2015 |
jiayl <jiayl@webrtc.org> |
Log to webrtc logging stream from java code.
Future log messages should all be sent to org.webrtc.Logging as well.
BUG=
Review URL: https://codereview.webrtc.org/1338033003
Cr-Commit-Position: refs/heads/master@{#9936}
pp/webrtc/java/android/org/webrtc/Camera2Enumerator.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/CameraEnumerator.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/GlShader.java
pp/webrtc/java/android/org/webrtc/GlUtil.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/Logging.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 1cb121dea478a4bb4f88e76cf92719e2853543cf |
14-Sep-2015 |
pbos <pbos@webrtc.org> |
Reset frame timestamp epoch for new capturers.
Incoming frames usually have an epoch of time since the capturer was
created or similar, not any fixed-time epoch. As such, setting a new
capturer resulted in delivering frames with older timestamps which
caused these frames to be dropped before encoding.
BUG=webrtc:4994
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345473002
Cr-Commit-Position: refs/heads/master@{#9934}
edia/base/fakevideocapturer.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| ea06a58f4018d85510acd96cf0304cb413a800b4 |
14-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android video capture: Remove duplicated code and fix spelling mistakes
This CL does not contain any functional changes, it is purely nit fixes.
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1340923002 .
Cr-Commit-Position: refs/heads/master@{#9931}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
|
| 92068ee6838b20cc204c5604f5f61c0253f1ff52 |
11-Sep-2015 |
colin <colin@comoyo.com> |
Android: Guard against switching camera on stopped camera
It is possible that cameraThreadHandler is null upon calling
switchCamera(). This CL adds a guard against that.
Review URL: https://codereview.webrtc.org/1325333003
Cr-Commit-Position: refs/heads/master@{#9925}
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
|
| c06a716d5941e8c60229afdd8720cd9c45178374 |
11-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Add new renderer SurfaceViewRenderer
BUG=webrtc:4742,webrtc:4910
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1308223002 .
Cr-Commit-Position: refs/heads/master@{#9922}
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
ibjingle.gyp
|
| d12140a68efdcffa1c2c18f25149905e9dae1a9c |
10-Sep-2015 |
guoweis <guoweis@webrtc.org> |
Revert change which removes GICE.
There are still dependencies on this functionality.
TBR=pthatcher@webrtc.org
BUG=526399
Review URL: https://codereview.webrtc.org/1336553003
Cr-Commit-Position: refs/heads/master@{#9920}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| fab882b193fa44f4c83c76fcbf67683b179978cf |
10-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove obsolete typingmonitor.cc/.h files.
To be committed once https://codereview.webrtc.org/1327033002/ has propagated to Chromium, and Chromium's libjingle.gyp has been updated.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1308663005
Cr-Commit-Position: refs/heads/master@{#9919}
ession/media/typingmonitor.cc
ession/media/typingmonitor.h
|
| 1dd98f321920c1442dd5b3f791ea0fca133c2756 |
10-Sep-2015 |
solenberg <solenberg@webrtc.org> |
- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1311533009
Cr-Commit-Position: refs/heads/master@{#9915}
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| 9a78d22822880884f9fa495e4cbe33f5224296c4 |
10-Sep-2015 |
tommi <tommi@webrtc.org> |
Revert of Consolidate constructormagic macros with Chromium version and remove Chromium override. (patchset #4 id:60001 of https://codereview.webrtc.org/1316363005/ )
Reason for revert:
Had to revert since FYI bots stopped compiling. Example failure:
[94/9470] CXX obj\third_party\webrtc\modules\video_processing\main\source\video_processing_sse2.content_analysis_sse2.obj
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\video_coding\codecs\h264\webrtc_h264.h264.obj.rsp /c ..\..\third_party\webrtc\modules\video_coding\codecs\h264\h264.cc /Foobj\third_party\webrtc\modules\video_coding\codecs\h264\webrtc_h264.h264.obj /Fdobj\third_party\webrtc\modules\webrtc_h264.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\base\rtc_base_approved.bitbuffer.obj.rsp /c ..\..\third_party\webrtc\base\bitbuffer.cc /Foobj\third_party\webrtc\base\rtc_base_approved.bitbuffer.obj /Fdobj\third_party\webrtc\base\rtc_base_approved.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\audio_processing\logging\audio_processing.aec_logging_file_handling.obj.rsp /c ..\..\third_party\webrtc\modules\audio_processing\logging\aec_logging_file_handling.cc /Foobj\third_party\webrtc\modules\audio_processing\logging\audio_processing.aec_logging_file_handling.obj /Fdobj\third_party\webrtc\modules\audio_processing.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\audio_processing\beamformer\audio_processing.nonlinear_beamformer.obj.rsp /c ..\..\third_party\webrtc\modules\audio_processing\beamformer\nonlinear_beamformer.cc /Foobj\third_party\webrtc\modules\audio_processing\beamformer\audio_processing.nonlinear_beamformer.obj /Fdobj\third_party\webrtc\modules\audio_processing.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
Original issue's description:
> Consolidate constructormagic macros with Chromium version and remove Chromium override.
>
> Part of work removing dependency on Chromium's base.
>
> Only adds "= delete". From https://codereview.chromium.org/1151443003 :
> "This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
>
> In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
>
> BUG=chromium:468375 (in particular comment #37)
> NOTRY=true
>
> Committed: https://crrev.com/0de8ff488d92e0bc6b7b65662898ff5e955cda93
> Cr-Commit-Position: refs/heads/master@{#9913}
TBR=andrew@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:468375 (in particular comment #37)
Review URL: https://codereview.webrtc.org/1330283002
Cr-Commit-Position: refs/heads/master@{#9914}
pp/webrtc/videosource.cc
|
| 0de8ff488d92e0bc6b7b65662898ff5e955cda93 |
10-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
BUG=chromium:468375 (in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1316363005
Cr-Commit-Position: refs/heads/master@{#9913}
pp/webrtc/videosource.cc
|
| c2db810b8958588771282634d00b7e3954c9f5ab |
09-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Remove VideoRendererInterface::CanApplyRotation()
All implementations handle rotation now, both internally in WebRTC and externally in Chromium.
R=glaznev@webrtc.org, guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1313753003 .
Cr-Commit-Position: refs/heads/master@{#9911}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
|
| f6901b06b87d5cc056ab222d68348763f1ed9544 |
09-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Remove NullVideoFrame
This class is not used.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1309743008 .
Cr-Commit-Position: refs/heads/master@{#9910}
ibjingle_tests.gyp
edia/base/nullvideoframe.h
|
| 8ce0bd54e9b603750cdc2246add1388b046f2182 |
09-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android video rendering: Fix texture matrix multiplication order
BUG=webrtc:4968, webrtc:4742
R=hbos@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1314163008 .
Cr-Commit-Position: refs/heads/master@{#9909}
pp/webrtc/java/android/org/webrtc/RendererCommon.java
|
| 2feafdb742226f57588d9c95bc25b2202166688f |
09-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Enable automatic resizing for RTX-enabled senders.
These were accidentally disabled due to checking ssrcs_.size() (which
includes RTX SSRCs) instead of rtp.ssrcs.size() to determine whether a
stream is simulcast or not.
BUG=webrtc:4965
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1318193003 .
Cr-Commit-Position: refs/heads/master@{#9907}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 529528cc36d2c34f886e34165a1635285db11b8a |
09-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android video rendering: Apply SurfaceTexture.getTransformationMatrix()
This CL applies the transformation matrix instead of assuming it is always a vertical flip.
BUG=webrtc:4968,webrtc:4742
R=hbos@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1318153007 .
Cr-Commit-Position: refs/heads/master@{#9905}
pp/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 66f43392a31ac566565e910246ef496fcbbafb04 |
09-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove [Voice|Video]MediaChannel::GetOptions().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1324853003
Cr-Commit-Position: refs/heads/master@{#9904}
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel_unittest.cc
|
| b04965ccf83c2bc6e2758abab9bea0c18551a54c |
09-Sep-2015 |
ivoc <ivoc@webrtc.org> |
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 7764973e1d5f8afaddab981cefb76b25477d8d94 |
08-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Add magjed@ as owner for talk/app/webrtc/androidtests/ and talk/app/webrtc/java/jni/
magjed@ has done a lot of work in these folders.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1314123004 .
Cr-Commit-Position: refs/heads/master@{#9896}
pp/webrtc/androidtests/OWNERS
pp/webrtc/java/jni/OWNERS
|
| 68786d20400f1f3744ad83549325665c18ea9e5b |
08-Sep-2015 |
stefan <stefan@webrtc.org> |
Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvoiceengine.cc
|
| e5269747595864eedd604f153df5d7bcbe1b475a |
08-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1325263002
Cr-Commit-Position: refs/heads/master@{#9891}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| f3ecdb981c172cdfafbe92c939eb25ddcc1ae96e |
08-Sep-2015 |
Henrik Boström <hbos@webrtc.org> |
Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1304043008 .
Cr-Commit-Position: refs/heads/master@{#9885}
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
|
| 8006f0759246407261b95c792f4febf3906415dc |
08-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove unused TypingMonitor class.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1327033002
Cr-Commit-Position: refs/heads/master@{#9884}
ibjingle.gyp
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/typingmonitor.cc
ession/media/typingmonitor.h
ession/media/typingmonitor_unittest.cc
|
| e9ad18b6e1606351b1a4187de52cf5ec6f9975bb |
08-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove obsolete soundclip.cc/.h files.
BUG=
Review URL: https://codereview.webrtc.org/1305033003
Cr-Commit-Position: refs/heads/master@{#9879}
ession/media/soundclip.cc
ession/media/soundclip.h
|
| 1c7d48d431e098ba42fa6bd9f1cfe69a703edee5 |
08-Sep-2015 |
Åsa Persson <asapersson@webrtc.org> |
Let max default bitrate depend on resolution when configuring one video stream (was previously always 2Mbps).
Is now set to:
<= 320x240: 600kbps
<= 640x480: 1.7Mbps
<= 960x540: 2Mbps
> 960x540: 2.5Mbps
For QVGA and VGA, the qp was around 10 at the selected thresholds when running some tests. The change in qp declined above the selected bitrates.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1297373003 .
Cr-Commit-Position: refs/heads/master@{#9878}
edia/webrtc/webrtcvideoengine2.cc
|
| 86d907cffda803ee34ee68f9833c1980d1b9f7a6 |
07-Sep-2015 |
henrika <henrika@webrtc.org> |
Refactor the AudioDevice for iOS and improve the performance and stability
This CL contains major modifications of the audio output parts for WebRTC on iOS:
- general code cleanup
- improves thread handling (added thread checks, remove critical section, atomic ops etc.)
- reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-)
- improves selection of audio parameters on iOS
- reduces complexity by removing complex and redundant delay estimates
- now instead uses fixed delay estimates if for some reason the SW EAC must be used
- adds AudioFineBuffer to compensate for differences in native output buffer size and
the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for
this class (the old code was buggy and we have several issue reports of crashes related to it)
Similar improvements will be done for the recording sid as well in a separate CL.
I will also add support for 48kHz in an upcoming CL since that will improve Opus performance.
BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212
TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice*
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1254883002 .
Cr-Commit-Position: refs/heads/master@{#9875}
edia/webrtc/webrtcvoiceengine.cc
|
| 7b38f6937087fed7011818a2692cd7f55c580813 |
07-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Add placeholder files for talk/app/webrtc/mediacontroller.cc/.h to be able to update Chrome's libjingle.gyp before the MediaController implementation CL is submitted.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1308543007
Cr-Commit-Position: refs/heads/master@{#9872}
pp/webrtc/mediacontroller.cc
pp/webrtc/mediacontroller.h
|
| bb741b3afa23ec59c1948841f2de71f422245564 |
07-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove GetOutputScaling from VoiceMediaChannel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1331443003
Cr-Commit-Position: refs/heads/master@{#9870}
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 0ab8ca83650899e50a42a9064aace2aaa5595c52 |
07-Sep-2015 |
phoglund <phoglund@webrtc.org> |
Remove x11 from libjingle_media
This generates incorrect -lX11 with use_x11=0 in our other build system,
which causes the standalone libjingle_media target to not build.
This patch should fix that. I could remove -lX11 completely, and
libjingle still links fine. So does Chrome if I do the corresponding
change there, so I think this change is safe to make.
BUG=None
Review URL: https://codereview.webrtc.org/1306243013
Cr-Commit-Position: refs/heads/master@{#9869}
ibjingle.gyp
|
| 9eb1365939683cc5462a5359344148efb7d84f97 |
05-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ )
Reason for revert:
It looks like this broke the FYI bots. I tried updating libjingle_nacl.gyp, but the IOS build still failed because in Chrome it's configured to use NSS. See https://codereview.chromium.org/1316863012/.
Original issue's description:
> purge nss files and dependencies
>
> BUG=webrtc:4497
>
> Committed: https://crrev.com/5647a2cf3db888195c928a1259d98f72f6ecbc15
> Cr-Commit-Position: refs/heads/master@{#9862}
TBR=tommi@webrtc.org,kjellander@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1311843006
Cr-Commit-Position: refs/heads/master@{#9867}
edia/sctp/sctpdataengine_unittest.cc
|
| 3cc834ae8628ef042497d8effe4bd223235bcd28 |
05-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Add more IceCandidatePairType for host-host CandidatePair
This is to help to differentiate endpoints which are behind NAT or on the public internet.
BUG=520101
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1328453003 .
Cr-Commit-Position: refs/heads/master@{#9864}
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| 5647a2cf3db888195c928a1259d98f72f6ecbc15 |
04-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
purge nss files and dependencies
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1313233005
Cr-Commit-Position: refs/heads/master@{#9862}
edia/sctp/sctpdataengine_unittest.cc
|
| e7a0de773a12a8d6efee3127582659e90feb1c4e |
04-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
CameraEnumerationAndroid: Add getSupportedFormats() implementation using android.hardware.camera2
Enumerating using android.hardware.camera2 is 10x faster than enumerating using android.hardware.camera, but they don't list exactly the same formats. android.hardware.camera2 support higher resolutions for some cameras, and also different framerates.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1321893003 .
Cr-Commit-Position: refs/heads/master@{#9861}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/Camera2Enumerator.java
pp/webrtc/java/jni/classreferenceholder.cc
ibjingle.gyp
|
| 47d78cc8ad54baabc9042c2b848ae3afd9b80d2e |
04-Sep-2015 |
sophiechang <sophiechang@chromium.org> |
Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder
BUG=
Review URL: https://codereview.webrtc.org/1263663005
Cr-Commit-Position: refs/heads/master@{#9853}
edia/webrtc/webrtcvideoengine2.cc
|
| dfbe679dedfa7048c4951138d192c07d65e268cc |
04-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Cleanup: Remove duplicated functions
IncrementCounter has been replaced by IncrementEnumCounter. Since the code has been rolled into Chromium, time to clean this up.
R=pthatcher@chromium.org
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1312763013 .
Cr-Commit-Position: refs/heads/master@{#9852}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| 658910cc3cb54705672c28fffedba4e982fa3989 |
03-Sep-2015 |
stefan <stefan@webrtc.org> |
Revert "Speculative revert of "- Move test cases for more natural ordering.""
Did not resolve the build bot issue.
This reverts commit 02d283a6ff5364d94aa88f5f5df4cfd3a5411346.
BUG=
TBR=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1324123002
Cr-Commit-Position: refs/heads/master@{#9849}
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 7afc12fe91e97a3d68de3768a73f3604e5651504 |
03-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoRendererGui: Move to async rendering and remove no longer needed code
BUG=webrtc:4742, webrtc:4909
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1321853003 .
Cr-Commit-Position: refs/heads/master@{#9847}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/java/android/org/webrtc/GlRectDrawer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| 1a591ddc7e6529e57a27f4a8f133ddd14a7ead16 |
02-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android GlUtil: Add helper functions generateTexture/deleteTexture
The purpose with this CL is to remove some code bloat. A subtle change is that GL_TEXTURE_MIN_FILTER in MediaCodecVideoDecoder is changed from GL_NEAREST to GL_LINEAR. This may lead to slightly worse performance when the decoded video is rendered minified, but with better visual quality. After reading https://crbug.com/351458 and the fix https://codereview.chromium.org/713603002 I think this is the right choice.
BUG=webrtc:4742
R=hbos@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1303373005 .
Cr-Commit-Position: refs/heads/master@{#9845}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/java/android/org/webrtc/GlUtil.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| ed4224fbdaca23a9ba593d07d6809a0943f73528 |
02-Sep-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android GlRectDrawer: Add fragment shader for RGB(A) textures
Add third shader type for RGB(A) and refactor according to the Rule of three.
BUG=webrtc:4742
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1311093005 .
Cr-Commit-Position: refs/heads/master@{#9843}
pp/webrtc/java/android/org/webrtc/GlRectDrawer.java
|
| e63d2a1c627566682512ca73c944c8a46e931e90 |
02-Sep-2015 |
Jiayang Liu <jiayl@chromium.org> |
Add JNI/java wrapper for the file rotating logging class.
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1309073004 .
Cr-Commit-Position: refs/heads/master@{#9840}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/CallSessionFileRotatingLogSink.java
pp/webrtc/java/src/org/webrtc/Logging.java
ibjingle.gyp
|
| 4d2f4d1c6997b4218774cac10527e1cdb88f20d9 |
02-Sep-2015 |
Alex Glaznev <glaznev@google.com> |
- Make shared EGL context used for HW video decoding member
of decoder factory class.
- Add new Peer connection factory method to initialize shared
EGL context.
This provides an option to use single peer connection factory
in the application and create peer connections from the same
factory and reinitialize shared EGL context for video
decoding HW acceleration.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1304063011 .
Cr-Commit-Position: refs/heads/master@{#9838}
pp/webrtc/androidtests/src/org/webrtc/PeerConnectionAndroidTest.java
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediadecoder_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 97579a4e122ad42592b5d7da9475e128da63a948 |
01-Sep-2015 |
glaznev <glaznev@webrtc.org> |
Add option to enable ECDSA key for Java API.
Review URL: https://codereview.webrtc.org/1312293003
Cr-Commit-Position: refs/heads/master@{#9835}
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
|
| eebc0996bfe99b66bdf6bec7dfc0fcd9bcb2ce06 |
01-Sep-2015 |
magjed <magjed@webrtc.org> |
Add magjed@ as owner for talk/app/webrtc/java/android/org/webrtc/
magjed@ has done a lot of work in this folder.
Review URL: https://codereview.webrtc.org/1322133002
Cr-Commit-Position: refs/heads/master@{#9834}
pp/webrtc/java/android/org/webrtc/OWNERS
|
| 194cceadae19a105074d4998408f06186ed43e42 |
01-Sep-2015 |
Alex Glaznev <glaznev@google.com> |
Do not use HW H.264 encoder on Nexus 7.
H.264 HW encoder on some Nexus 7 models have
poor bitrate control.
R=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1311893009 .
Cr-Commit-Position: refs/heads/master@{#9833}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 4edc39c5692ab01286d0c4a4911bf6b705032b6f |
01-Sep-2015 |
honghaiz <honghaiz@webrtc.org> |
Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.
BUG= 4901
Review URL: https://codereview.webrtc.org/1315503003
Cr-Commit-Position: refs/heads/master@{#9832}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/objc/RTCPeerConnectionInterface.mm
pp/webrtc/objc/public/RTCPeerConnectionInterface.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
pp/webrtc/webrtcsession.cc
|
| 02d283a6ff5364d94aa88f5f5df4cfd3a5411346 |
01-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
Speculative revert of "- Move test cases for more natural ordering."
This reverts commit c20a5dc9305b988ca173cd63e606124b02e6d54c.
BUG=webrtc:4959
R=solenberg@webrtc.org
TBR=solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1309313008 .
Cr-Commit-Position: refs/heads/master@{#9829}
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| c252dabbd6bc98b0f725d221e58431128d4eed07 |
31-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
CameraEnumerationAndroid: Make getSupportedFormats() an interface
Enumerating camera capabilities in the deprecated android.hardware.Camera interface is really slow because of the need to open and release the camera. By making getSupportedFormats() an interface, we allow apps the opportunity to inject their own implementation, such as storing the supported formats offline in the device's internal storage. It will also be possible to add an implementation of getSupportedFormats() using the new android.hardware.Camera2 interface in a follow-up CL.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1321903002 .
Cr-Commit-Position: refs/heads/master@{#9819}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/CameraEnumerator.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/jni/classreferenceholder.cc
ibjingle.gyp
|
| c20a5dc9305b988ca173cd63e606124b02e6d54c |
31-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
- Move test cases for more natural ordering.
- Get rid of the CoInitialize tests for WVoE/WViE.
BUG=webrtc:4690
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1319163002 .
Cr-Commit-Position: refs/heads/master@{#9817}
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 3a14bf311f366602ebc72314ca8906be61a70da4 |
31-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers.
Updates TransportDescriptionFactory, calls and unittests.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1311903004 .
Cr-Commit-Position: refs/heads/master@{#9815}
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession_unittest.cc
|
| a6cba3ab5c899339d577adf1824e0e007c12863e |
29-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Java VideoRenderer.Callbacks: Make renderFrame() interface asynchronous
This CL makes the Java render interface asynchronous by requiring every call to renderFrame() to be followed by an explicit renderFrameDone() call. In JNI, this is implemented with cricket::VideoFrame::Copy() before calling renderFrame(), and a corresponding call to delete in renderFrameDone(). This CL is primarily done to prepare for a new renderer implementation.
BUG=webrtc:4742, webrtc:4909
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1313563002 .
Cr-Commit-Position: refs/heads/master@{#9814}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
|
| e8386d21992b6683b6fadd315ea631875b4256fb |
28-Aug-2015 |
Lally Singh <lally@google.com> |
Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there.
BUG=https://code.google.com/p/webrtc/issues/detail?id=4468
R=pthatcher@chromium.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1304063006 .
Cr-Commit-Position: refs/heads/master@{#9812}
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
|
| 79de90b1100853c78594705c272528eea7b706d5 |
28-Aug-2015 |
Alex Glaznev <glaznev@google.com> |
Do not explicitly delete OpenGL shaders in VideoRendererGui.
This is handled by Android itself and may result in GL errors
when trying to release shaders when Activity is destroyed.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1322703004 .
Cr-Commit-Position: refs/heads/master@{#9811}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| f42376c60111edba6f29060bf3dd79e75d8dbb97 |
28-Aug-2015 |
pbos <pbos@webrtc.org> |
Wire up currently-received video codec to stats.
BUG=webrtc:1844, webrtc:4808
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1315413002
Cr-Commit-Position: refs/heads/master@{#9810}
pp/webrtc/statscollector.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 6813ec84fbe813e259e7b2f7d65f6d79e221e2ab |
28-Aug-2015 |
magjed <magjed@webrtc.org> |
VideoCapturerAndroid: Move to android folder and split out camera enumeration into separate file
Pure code move of:
talk/app/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
into:
talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1323453002
Cr-Commit-Position: refs/heads/master@{#9809}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
pp/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
ibjingle.gyp
|
| 4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d |
28-Aug-2015 |
solenberg <solenberg@webrtc.org> |
Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1273363005
Cr-Commit-Position: refs/heads/master@{#9807}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 6480d03f176888999b56a2fa09ddf368f5ff5913 |
28-Aug-2015 |
phoglund <phoglund@webrtc.org> |
Make jni_helpers build on arm32.
BUG=None
Review URL: https://codereview.webrtc.org/1311753002
Cr-Commit-Position: refs/heads/master@{#9806}
pp/webrtc/java/jni/jni_helpers.cc
|
| 6ec1f921b1186127467393dc82c0a786e0de4e2b |
28-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
AndroidVideoCapturer: Delegate framerate choice to VideoCapturerAndroid.java
webrtc::VideoSource resolves the kMaxFrameRate constraint by capping the desired framerate to kMaxFrameRate. That framerate is then passed into cricket::VideoCapturer::GetBestCaptureFormat(). The default implementation will choose a format from the supported formats list. Instead, we should override this function in AndroidVideoCapturer to give VideoCapturerAndroid.java the opportunity to choose a suitable framerate range.
BUG=webrtc:4938
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1308953004 .
Cr-Commit-Position: refs/heads/master@{#9805}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/videosource.cc
|
| 1c3dd38cb819733fa3f558063d4b0c135c5be6e7 |
27-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Fix memory leak for remote MediaStream
BUG=webrtc:4892
R=glaznev@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1308733004 .
Cr-Commit-Position: refs/heads/master@{#9797}
pp/webrtc/java/jni/jni_helpers.cc
pp/webrtc/java/jni/jni_helpers.h
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 7391881f9762ccadeeb0249560b33cf2bcfaf7f9 |
27-Aug-2015 |
tommi <tommi@webrtc.org> |
Revert of Added send-thresholding and fixed text packet dumping. (patchset #4 id:160001 of https://codereview.webrtc.org/1266033005/ )
Reason for revert:
The CL adds a global variable.
Original issue's description:
> Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there.
>
> BUG=https://code.google.com/p/webrtc/issues/detail?id=4468
> R=pthatcher@chromium.org, pthatcher@webrtc.org
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/d838d2791979bb50f464a61c557d55c6a324621e
TBR=pthatcher@webrtc.org,bemasc@webrtc.org,pthatcher@chromium.org,thakis@chromium.org,lally@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=https://code.google.com/p/webrtc/issues/detail?id=4468
Review URL: https://codereview.webrtc.org/1315923003
Cr-Commit-Position: refs/heads/master@{#9796}
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
|
| fdac516510a2bd5d57b0786fbd49e2a6b9aeed2f |
27-Aug-2015 |
noahric <noahric@chromium.org> |
Disallow simulcast for H.264.
BUG=
Review URL: https://codereview.webrtc.org/1291673006
Cr-Commit-Position: refs/heads/master@{#9795}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| d82819892a382899a82ced756a9922a84ca9ca98 |
27-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer.
Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.
This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1312643004 .
Cr-Commit-Position: refs/heads/master@{#9794}
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ession/media/channel_unittest.cc
|
| c47a01d6477da02fddeca23a4efb32f5764af128 |
27-Aug-2015 |
Alex Glaznev <glaznev@google.com> |
Fix AppRTCDemo crash when room is connected after PC is destroyed.
Also move VideoRendererGui.dispose() to the section with public API.
BUG=4909
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1312523004 .
Cr-Commit-Position: refs/heads/master@{#9792}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| d838d2791979bb50f464a61c557d55c6a324621e |
26-Aug-2015 |
Lally Singh <lally@google.com> |
Added send-thresholding and fixed text packet dumping. Also a little squelch for the over-max-MTU log spam we see in there.
BUG=https://code.google.com/p/webrtc/issues/detail?id=4468
R=pthatcher@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1266033005 .
Cr-Commit-Position: refs/heads/master@{#9788}
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
|
| 3318f984cd7f51d24da4726665c05f5f06f82e6d |
26-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoFrameBuffer: Make non-const data access explicit
VideoFrameBuffer currently has two overloaded data() functions for pixel access, one for const and one for non-const. Unfortunately, it will default to the non-const version, even when 'const scoped_refptr<VideoFrameBuffer>&' is used. This is a problem, because many subclasses use RTC_NOTREACHED() in the non-const version.
This CL makes the non-const version of data() explicit with a different, longer function name MutableData().
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1304143003 .
Cr-Commit-Position: refs/heads/master@{#9787}
edia/webrtc/webrtcvideoframe.cc
|
| 85ad62b87760213a1a453051d833c8c40e82d9bd |
26-Aug-2015 |
Noah Richards <noahric@chromium.org> |
Remove per-frame captured frame logging.
It's a little too verbose :)
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1302173004 .
Cr-Commit-Position: refs/heads/master@{#9786}
edia/webrtc/webrtcvideoengine2.cc
|
| af9fb218864b8cb4cccd32280b68dd1b34cb2213 |
26-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
- Use C++11 loops in WebRtcVoiceMediaEngine/Channel.
- Pull out part of WebRtcVoiceMediaChannel::SetRecvCodecs() into WebRtcVoiceMediaChannel::SetRecvCodecsInternal().
BUG=webrtc:4690
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1291343002 .
Cr-Commit-Position: refs/heads/master@{#9785}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| c464f504dcb40ad40b5258875493f12783bd5fda |
25-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
AndroidVideoCapturerJni: Fix threading issues
The primary fix in this CL is to remove the dangling |thread_| pointer in AndroidVideoCapturerJni. That thread is not safe to use after Stop() has been called. Even after Stop() has been called, we must still be able to return late frames to Java in order to not leak them, so that path has been made thread safe instead. To make sure that we always return frames, the Java frame should be wrapped in a scoped_refptr as quickly as possible, so this CL moves the wrapping from AndroidVideoCapturer to AndroidVideoCapturerJni. This also removes the need for the interface function AndroidVideoCapturerDelegate::ReturnBuffer().
Some other minor changes are:
* Remove |valid_global_refs_| and all logic related to that. Now that rtc::Bind() captures method objects as scoped_refptr, the destructor of AndroidVideoCapturerJni will not be called before all frames are returned.
* Remove global ref |j_frame_observer_|. No need for this, we don’t call it and it is kept alive with standard Java memory management.
* Add helper function ShallowCenterCrop() for VideoFrameBuffers. This functionality already exists in the constructor of WrappedI420Buffer, but it’s more convenient to have it as a separate function.
BUG=webrtc:4742,webrtc:4909
R=glaznev@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1307973002 .
Cr-Commit-Position: refs/heads/master@{#9784}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| c464b409be3c19ee3b0a78b5d60053acef00d55b |
25-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android RendererCommon: Add unittests for getTextureMatrix()
BUG=webrtc:4742
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1315843002 .
Cr-Commit-Position: refs/heads/master@{#9783}
pp/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java
|
| 7230a21d66eb7d71f35b7362d23dcd5199635e56 |
25-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android RendererCommon: Add unittests for getDisplaySize()
BUG=webrtc:4742
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1313513002 .
Cr-Commit-Position: refs/heads/master@{#9777}
pp/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
|
| 87713d0fe6fb9c86abe501bdf3d26ef4287ee617 |
25-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
RTCCertificates added to RTCConfiguration, used by WebRtcSession/-DescriptionFactory.
This CL allows you to, having generated one or more RTCCertificates, supply them to RTCConfiguration for CreatePeerConnection use. This means an SSLIdentity does not have to be generated with a DtlsIdentityStore[Interface/Impl] as part of the CreatePeerConnection steps because the certificate contains all the necessary information.
To create an RTCCertificate you have to do the identity generation yourself though. But you could reuse the same RTCCertificate for multiple connections.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1288033009 .
Cr-Commit-Position: refs/heads/master@{#9774}
pp/webrtc/peerconnectioninterface.h
pp/webrtc/test/fakedtlsidentitystore.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| 7ef9d9104d2143fdd775f7d77bcb698b4d919d59 |
25-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Remove VideoRenderer.Callbacks.canApplyRotation()
The only real implementation of VideoRenderer.Callbacks, VideoRendererGui, can always apply rotation. We don't need this in the interface.
BUG=webrtc:4145
R=glaznev@webrtc.org, guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1306073003 .
Cr-Commit-Position: refs/heads/master@{#9772}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreaminterface.h
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| 2159b89fa2cb55beeef38f72bd45e217f3d33d4e |
22-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.
Original CL: https://codereview.webrtc.org/1263663002/
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1303393002 .
Cr-Commit-Position: refs/heads/master@{#9761}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| ea1012b2a41b1b56fe7366792f10390639d82495 |
21-Aug-2015 |
guoweis <guoweis@webrtc.org> |
address comments from https://codereview.webrtc.org/1277263002/
TBR=juberti@webrtc.org,pthather@webrtc.org
Review URL: https://codereview.webrtc.org/1305113002
Cr-Commit-Position: refs/heads/master@{#9757}
pp/webrtc/fakemetricsobserver.cc
|
| 5bdafd44c86ee46bd7e040f19828324583418b33 |
21-Aug-2015 |
minyuel <minyue@webrtc.org> |
Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.
Original code review see
https://codereview.webrtc.org/1291363005
The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137
TBR=pthatcher@webrtc.org,
BUG=
Review URL: https://codereview.webrtc.org/1308753003 .
Cr-Commit-Position: refs/heads/master@{#9756}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| c232096eba8aa96b7dcbf52a1e956713c07e9972 |
21-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Remove cricket::VideoProcessor and AddVideoProcessor() functionality
This functionality is not used internally in WebRTC. Also, it's not safe, because the frame is supposed to be read-only, and it will likely not work for texture frames.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1296113002 .
Cr-Commit-Position: refs/heads/master@{#9753}
ibjingle.gyp
edia/base/capturemanager.cc
edia/base/capturemanager.h
edia/base/capturemanager_unittest.cc
edia/base/capturerenderadapter.cc
edia/base/fakemediaengine.h
edia/base/fakemediaprocessor.h
edia/base/mediaengine.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoprocessor.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| bfab5cbc33f21eccdd8e320ca44201ad6711f542 |
21-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Fix some minor errors with the voice engine caused by the refactor CL https://codereview.webrtc.org/1229283003/.
R=deadbeef@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1284693003 .
Cr-Commit-Position: refs/heads/master@{#9750}
ession/media/channel.cc
|
| a5b273a635b9876f88430934de19a883a1fb5728 |
21-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing problems with RTP extension ID conflict resolution
If the same extension URI is used for both audio and video (such as
abs-send-time), we should be able to re-use the same ID. A conflict
only exists if two different URIs are attempting to use the same ID.
Review URL: https://codereview.webrtc.org/1286273003
Cr-Commit-Position: refs/heads/master@{#9749}
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 874ca3af5b163e1b3fd8802171e44ee252557842 |
21-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
Don't do reconfiguration if recv codec order/preference changes
Adding 'ReceiveCodecsHaveChanged' method that will determine if codecs
HAVE changed, irrespective of order and preference.
Review URL: https://codereview.webrtc.org/1291763003
Cr-Commit-Position: refs/heads/master@{#9748}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| fe3bc9d5aeffed8bbfb34c330d8b991abd1a1aba |
20-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Relanding "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
Migrated from https://codereview.webrtc.org/1275703006/ which causes test failures for android. On android, loopback interface was used as local interface to generate candidates. Add a test case to make sure this won't be broken in the future.
Also observed some failures under content_browsertests in chromium.fyi bot but can't repro locally. Might just be temporary test issue.
BUG=webrtc:4517
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1299333003 .
Cr-Commit-Position: refs/heads/master@{#9746}
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| ff020c01ca8ad72e513700315ebb6ff16afffd22 |
20-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android: Move common functions from VideoRendererGui to new RendererCommon file
This is primarily done to prepare for a new renderer implementation.
BUG=webrtc:4742
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1298673002 .
Cr-Commit-Position: refs/heads/master@{#9742}
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
ibjingle.gyp
|
| d476b955046aee5d5f750111b567f5ed51bf85b8 |
20-Aug-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android EglBase: Add helper functions to query the surface size
BUG=webrtc:4742
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1299543004 .
Cr-Commit-Position: refs/heads/master@{#9739}
pp/webrtc/java/android/org/webrtc/EglBase.java
|
| 081f34b564e1a26ffbbe9515eba1fef7c736fdde |
20-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1291363005 .
Cr-Commit-Position: refs/heads/master@{#9738}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| 3d564c10157d7de1d2d4236f4e2a13ff1363d52b |
20-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Add instrumentation to track the IceEndpointType.
The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.
BUG=webrtc:4918
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1277263002 .
Cr-Commit-Position: refs/heads/master@{#9737}
pp/webrtc/fakemetricsobserver.cc
pp/webrtc/fakemetricsobserver.h
pp/webrtc/objc/RTCEnumConverter.mm
pp/webrtc/objc/public/RTCTypes.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| 048e80cacab128280c37c3e0574873cde8544410 |
19-Aug-2015 |
tommi <tommi@webrtc.org> |
Revert of Revert "Remove CpuMonitor and related, unused, code." (patchset #1 id:1 of https://codereview.webrtc.org/1287913004/ )
Reason for revert:
(retrying with my webrtc account...)
The reason for reverting is: Re-landing the change that removes the CpuMonitor class after having fixed the build issue in Chromium..
Original issue's description:
> Revert "Remove CpuMonitor and related, unused, code."
>
> This reverts commit 1a24012680f25440aa1d117373df2af14cdc2fc1.
>
> TBR=tommi@webrtc.org,pthatcher@webrtc.org
> BUG=
>
> This breaks
> http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/20148/steps/compile/logs/stdio
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/a472e968c95fb14e63ec42f453551d0967573ea8
TBR=pthatcher@webrtc.org,guoweis@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1290033005
Cr-Commit-Position: refs/heads/master@{#9733}
edia/webrtc/webrtcvideoengine2.h
|
| a472e968c95fb14e63ec42f453551d0967573ea8 |
19-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Remove CpuMonitor and related, unused, code."
This reverts commit 1a24012680f25440aa1d117373df2af14cdc2fc1.
TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=
This breaks
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/20148/steps/compile/logs/stdio
Review URL: https://codereview.webrtc.org/1287913004 .
Cr-Commit-Position: refs/heads/master@{#9730}
edia/webrtc/webrtcvideoengine2.h
|
| 370c8848ad38d54457a960e0ebe94f8adf370e23 |
19-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied."
This reverts commit 0a2955f227666efd87b2a303a69c083ef801c528.
Revert "In the past, P2PPortAllocator.enable_multiple_routes is the indicator whether we should bind to the any address. It's easy to translate that into a port allocator flag in P2PPortAllocator's ctor. Going forward, we have to depend on an asynchronous permission check to determine whether gathering local address is allowed or not, hence the current way of passing it through constructor approach won't work any more. The asynchronous check will trigger SignalNetowrksChanged so we could only check that inside DoAllocate."
This reverts commit ba9ab4cd8d2e8fbc068dc36b5e6f6331d7deeccf.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1288843003 .
Cr-Commit-Position: refs/heads/master@{#9729}
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| 1a24012680f25440aa1d117373df2af14cdc2fc1 |
18-Aug-2015 |
Tommi <tommi@webrtc.org> |
Remove CpuMonitor and related, unused, code.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1298953002 .
Cr-Commit-Position: refs/heads/master@{#9727}
edia/webrtc/webrtcvideoengine2.h
|
| 0a2955f227666efd87b2a303a69c083ef801c528 |
18-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Generate localhost candidate when no STUN/TURN and portallocator has the right flag spefied.
BUG=webrtc:4517
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1275703006 .
Cr-Commit-Position: refs/heads/master@{#9726}
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| dbe5bd9ad58c9d53289ff678b2db54433ac63a07 |
17-Aug-2015 |
Nico Weber <thakis@chromium.org> |
Delete unused function SetSessionError.
https://webrtc-codereview.appspot.com/47589004/ remove the use.
BUG=505316
Originally reviewed at https://codereview.webrtc.org/1296103002/
TBR=sergeyu@chromium.org
Review URL: https://codereview.webrtc.org/1299703002 .
Cr-Commit-Position: refs/heads/master@{#9719}
ession/media/channel.cc
|
| b6d4ec418504fd947c6f96829c73180e9487e203 |
17-Aug-2015 |
Torbjorn Granlund <torbjorng@google.com> |
Support generation of EC keys using P256 curve and support ECDSA certs.
This CL started life here: https://webrtc-codereview.appspot.com/51189004
BUG=webrtc:4685, webrtc:4686
R=hbos@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1189583002 .
Cr-Commit-Position: refs/heads/master@{#9718}
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/channel_unittest.cc
|
| 55e9a7dc4ba922599d4fbcf25f13629ba210fc64 |
14-Aug-2015 |
Alex Glaznev <glaznev@google.com> |
Add Android VideoRendererGui events.
Add events to Android VideoRendererGui implementation to
optionally report first rendered frame and video frame
dimension changes.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1292293002 .
Cr-Commit-Position: refs/heads/master@{#9715}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 60d9b332a5391045439bfb6a3a5447973e3d5603 |
14-Aug-2015 |
ekmeyerson <ekmeyerson@webrtc.org> |
Integrate Intelligibility with APM
- Integrates intelligibility into audio_processing.
- Allows modification of reverse stream if intelligibility enabled.
- Makes intelligibility available in audioproc_float test.
- Adds reverse stream processing to audioproc_float.
- (removed) Makes intelligibility toggleable in real time in voe_cmd_test.
- Cleans up intelligibility construction, parameters, constants and dead code.
TBR=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1234463003
Cr-Commit-Position: refs/heads/master@{#9713}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 4c530dccb33be5a3089a47efd280b1585ec7afd2 |
14-Aug-2015 |
hbos <hbos@webrtc.org> |
Delete dummy dtlsidentityservice.[cc,h] files.
BUG=webrtc:4899
Review URL: https://codereview.webrtc.org/1284383005
Cr-Commit-Position: refs/heads/master@{#9711}
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
|
| d5031fcf928b615a1c9aa9f15e60d1c946e6d456 |
14-Aug-2015 |
magjed <magjed@webrtc.org> |
Android VideoRendererGui: Add dispose function
There is currently no way to dispose VideoRendererGui or VideoRendererGui.YuvImageRenderer. This CL adds functions to do so.
BUG=webrtc:4892
Review URL: https://codereview.webrtc.org/1273803002
Cr-Commit-Position: refs/heads/master@{#9710}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| af5c035e4348393c299f8cb687eebd3fa5b55ecb |
14-Aug-2015 |
magjed <magjed@webrtc.org> |
VideoCapturerAndroid: Release queued camera frames when stopCapture() is called
BUG=webrtc:4892
Review URL: https://codereview.webrtc.org/1285823002
Cr-Commit-Position: refs/heads/master@{#9709}
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| ee8c6d327357ecd2e17edede8d15f6e3893409a8 |
13-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
In PeerConnectionTestWrapper, put audio input on a separate thread.
This will prevent it from blocking network input when it falls behind,
which is happening when running with ThreadSanitizer.
BUG=webrtc:4663
Review URL: https://codereview.webrtc.org/1236023010
Cr-Commit-Position: refs/heads/master@{#9707}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
|
| c558af854f5f14be49652b684d3e2c328fb071d7 |
13-Aug-2015 |
hbos <hbos@webrtc.org> |
Removing DtlsIdentityService[Interface] which has been replaced by DtlsIdentityStore[Interface/Impl].
This is CL is part of an effort to land https://codereview.webrtc.org/1176383004 without breaking Chromium.
See bug for more information.
BUG=webrtc:4899
Review URL: https://codereview.webrtc.org/1282413002
Cr-Commit-Position: refs/heads/master@{#9705}
pp/webrtc/dtlsidentitystore.h
pp/webrtc/peerconnectioninterface.h
|
| e2a8be124458d77d0d3f30a8e33e0a1eede4a849 |
12-Aug-2015 |
magjed <magjed@webrtc.org> |
Revert of AppRTCDemo: Render each video in a separate SurfaceView (patchset #4 id:120001 of https://codereview.webrtc.org/1257043004/ )
Reason for revert:
AppRTCDemo often crashes in loopback mode and incorrect layout when connection is established
BUG=webrtc:4909,webrtc:4910
Original issue's description:
> AppRTCDemo: Render each video in a separate SurfaceView
>
> This CL introduces a new org.webrtc.VideoRenderer.Callbacks implementation called SurfaceViewRenderer that renders each video stream in its own SurfaceView. AppRTCDemo is updated to use this new rendering.
>
> This CL also does the following changes:
> * Make the VideoRenderer.Callbacks interface asynchronous and require that renderFrameDone() is called for every renderFrame(). In JNI, this is implemented with cricket::VideoFrame::Copy()/delete.
> * Make public static helper functions: convertScalingTypeToVisibleFraction(), getDisplaySize(), and getTextureMatrix().
> * Introduces new helper functions surfaceWidth()/surfaceHeight() in EGlBase that allows to query the surface size.
> * Introduce PercentFrameLayout that implements the percentage layout that is used by AppRTCDemo.
>
> BUG=webrtc:4742
>
> Committed: https://crrev.com/05bfbe47ef6bcc9ca731c0fa0d5cd15a4f21e93f
> Cr-Commit-Position: refs/heads/master@{#9699}
TBR=glaznev@webrtc.org,wzh@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1286133002
Cr-Commit-Position: refs/heads/master@{#9703}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
ibjingle.gyp
|
| d941b7609c64516f1b93a0909a0bc23c10d26173 |
12-Aug-2015 |
budnyjj <budnyjj@gmail.com> |
Fix distortions of remote stream with odd size dimensions
BUG=webrtc:4482
Review URL: https://codereview.webrtc.org/1280483003
Cr-Commit-Position: refs/heads/master@{#9702}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 8a2cd3d57da1eb0494aff3b14a7ff34fe97fd5ef |
11-Aug-2015 |
Alex Glaznev <glaznev@google.com> |
Revert H.264 HW encoder setting to CBR mode.
VBR mode does not work well on KK devices - bitrate
deviations from target are too large,
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1270403007 .
Cr-Commit-Position: refs/heads/master@{#9701}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 05bfbe47ef6bcc9ca731c0fa0d5cd15a4f21e93f |
11-Aug-2015 |
magjed <magjed@webrtc.org> |
AppRTCDemo: Render each video in a separate SurfaceView
This CL introduces a new org.webrtc.VideoRenderer.Callbacks implementation called SurfaceViewRenderer that renders each video stream in its own SurfaceView. AppRTCDemo is updated to use this new rendering.
This CL also does the following changes:
* Make the VideoRenderer.Callbacks interface asynchronous and require that renderFrameDone() is called for every renderFrame(). In JNI, this is implemented with cricket::VideoFrame::Copy()/delete.
* Make public static helper functions: convertScalingTypeToVisibleFraction(), getDisplaySize(), and getTextureMatrix().
* Introduces new helper functions surfaceWidth()/surfaceHeight() in EGlBase that allows to query the surface size.
* Introduce PercentFrameLayout that implements the percentage layout that is used by AppRTCDemo.
BUG=webrtc:4742
Review URL: https://codereview.webrtc.org/1257043004
Cr-Commit-Position: refs/heads/master@{#9699}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/RendererCommon.java
pp/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
ibjingle.gyp
|
| fa301809b698017455847f45cc7e0dfa1bdfed35 |
11-Aug-2015 |
pthatcher <pthatcher@webrtc.org> |
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| cc4ebadf0b5783fc46f1279534aa5e838945c5d7 |
11-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Empty dtlsidentityservice.h/cc files added, to be removed once chromium gyp files don't reference it.
This is CL is part of an effort to land https://codereview.webrtc.org/1176383004 without breaking Chromium.
See bug for more information.
BUG=webrtc:4899
TBR=tommi@webrtc.org,magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1276233006 .
Cr-Commit-Position: refs/heads/master@{#9697}
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
|
| 5e56c5927e097f095aef2e9f7be49fd3d59221e1 |
11-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface).
DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002
The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.
Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.
For more information about the steps being taken to land this without breaking Chromium, see referenced bug.
BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1176383004 .
Cr-Commit-Position: refs/heads/master@{#9696}
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakedtlsidentityservice.h
pp/webrtc/test/fakedtlsidentitystore.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ibjingle.gyp
ibjingle_tests.gyp
|
| 3449faa553ec94c52ef2d0949867befb60992c88 |
10-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1263663002 .
Cr-Commit-Position: refs/heads/master@{#9692}
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| c2ee2c86f905991a8cd05ee1f35bea105b41e4e0 |
08-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229283003 .
Cr-Commit-Position: refs/heads/master@{#9690}
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
ession/media/channel.cc
ession/media/channel.h
|
| 25c96d02cdd2460b378ab89e4b90b17a81bf0d4a |
07-Aug-2015 |
jbauch <jbauch@webrtc.org> |
Add thread checker to StatsCollection.
This CL makes sure the methods are always called on the correct thread.
Review URL: https://codereview.webrtc.org/1235263003
Cr-Commit-Position: refs/heads/master@{#9688}
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 0482dcc87376468be5f8a1a6d8f1d8a4f58267e8 |
07-Aug-2015 |
Alex Glaznev <glaznev@google.com> |
Enable HW H.264 decoding on Intel platforms.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1274133003 .
Cr-Commit-Position: refs/heads/master@{#9686}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| fcf8ece6ba1d170fc70a457d93b35eccb3074022 |
06-Aug-2015 |
magjed <magjed@webrtc.org> |
AndroidVideoCapturer: Return frames that have been dropped
Currently, we only return frames if CreateAliasedFrame() is called, which is not the case for dropped frames.
Review URL: https://codereview.webrtc.org/1268333005
Cr-Commit-Position: refs/heads/master@{#9683}
pp/webrtc/androidvideocapturer.cc
|
| a8736448970fedd82f051c6b2cc89185b755ddf3 |
06-Aug-2015 |
Donald E Curtis <decurtis@google.com> |
Move all the examples from the talk directory into the webrtc examples directory.
Significant changes:
- move the libjingle_examples.gyp file into webrtc directory.
- rename talk/examples/android to webrtc/examples/androidapp to avoid name conflicts.
- update paths in talk/libjingle_tests.gyp to point to webrtc directory for Objective-C test.
BUG=
R=pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1235563006 .
Cr-Commit-Position: refs/heads/master@{#9681}
xamples/OWNERS
xamples/android/AndroidManifest.xml
xamples/android/README
xamples/android/ant.properties
xamples/android/build.xml
xamples/android/project.properties
xamples/android/res/drawable-hdpi/disconnect.png
xamples/android/res/drawable-hdpi/ic_action_full_screen.png
xamples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-hdpi/ic_launcher.png
xamples/android/res/drawable-hdpi/ic_loopback_call.png
xamples/android/res/drawable-ldpi/disconnect.png
xamples/android/res/drawable-ldpi/ic_action_full_screen.png
xamples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-ldpi/ic_launcher.png
xamples/android/res/drawable-ldpi/ic_loopback_call.png
xamples/android/res/drawable-mdpi/disconnect.png
xamples/android/res/drawable-mdpi/ic_action_full_screen.png
xamples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-mdpi/ic_launcher.png
xamples/android/res/drawable-mdpi/ic_loopback_call.png
xamples/android/res/drawable-xhdpi/disconnect.png
xamples/android/res/drawable-xhdpi/ic_action_full_screen.png
xamples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-xhdpi/ic_launcher.png
xamples/android/res/drawable-xhdpi/ic_loopback_call.png
xamples/android/res/layout/activity_call.xml
xamples/android/res/layout/activity_connect.xml
xamples/android/res/layout/fragment_call.xml
xamples/android/res/layout/fragment_hud.xml
xamples/android/res/menu/connect_menu.xml
xamples/android/res/values-v17/styles.xml
xamples/android/res/values-v21/styles.xml
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCProximitySensor.java
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/CallFragment.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/CpuMonitor.java
xamples/android/src/org/appspot/apprtc/HudFragment.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/SettingsFragment.java
xamples/android/src/org/appspot/apprtc/UnhandledExceptionHandler.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/src/org/appspot/apprtc/util/AppRTCUtils.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/android/src/org/appspot/apprtc/util/LooperExecutor.java
xamples/android/third_party/autobanh/LICENSE
xamples/android/third_party/autobanh/LICENSE.md
xamples/android/third_party/autobanh/NOTICE
xamples/android/third_party/autobanh/autobanh.jar
xamples/androidtests/AndroidManifest.xml
xamples/androidtests/README
xamples/androidtests/ant.properties
xamples/androidtests/build.xml
xamples/androidtests/project.properties
xamples/androidtests/src/org/appspot/apprtc/test/LooperExecutorTest.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
xamples/objc/.clang-format
xamples/objc/AppRTCDemo/ARDAppClient+Internal.h
xamples/objc/AppRTCDemo/ARDAppClient.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.h
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/ARDCEODTURNClient.h
xamples/objc/AppRTCDemo/ARDCEODTURNClient.m
xamples/objc/AppRTCDemo/ARDJoinResponse+Internal.h
xamples/objc/AppRTCDemo/ARDJoinResponse.h
xamples/objc/AppRTCDemo/ARDJoinResponse.m
xamples/objc/AppRTCDemo/ARDMessageResponse+Internal.h
xamples/objc/AppRTCDemo/ARDMessageResponse.h
xamples/objc/AppRTCDemo/ARDMessageResponse.m
xamples/objc/AppRTCDemo/ARDRoomServerClient.h
xamples/objc/AppRTCDemo/ARDSDPUtils.h
xamples/objc/AppRTCDemo/ARDSDPUtils.m
xamples/objc/AppRTCDemo/ARDSignalingChannel.h
xamples/objc/AppRTCDemo/ARDSignalingMessage.h
xamples/objc/AppRTCDemo/ARDSignalingMessage.m
xamples/objc/AppRTCDemo/ARDTURNClient.h
xamples/objc/AppRTCDemo/ARDWebSocketChannel.h
xamples/objc/AppRTCDemo/ARDWebSocketChannel.m
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.h
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.m
xamples/objc/AppRTCDemo/RTCICEServer+JSON.h
xamples/objc/AppRTCDemo/RTCICEServer+JSON.m
xamples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h
xamples/objc/AppRTCDemo/RTCMediaConstraints+JSON.m
xamples/objc/AppRTCDemo/RTCSessionDescription+JSON.h
xamples/objc/AppRTCDemo/RTCSessionDescription+JSON.m
xamples/objc/AppRTCDemo/common/ARDUtilities.h
xamples/objc/AppRTCDemo/common/ARDUtilities.m
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.h
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.m
xamples/objc/AppRTCDemo/ios/ARDMainView.h
xamples/objc/AppRTCDemo/ios/ARDMainView.m
xamples/objc/AppRTCDemo/ios/ARDMainViewController.h
xamples/objc/AppRTCDemo/ios/ARDMainViewController.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
xamples/objc/AppRTCDemo/ios/AppRTCDemo-Prefix.pch
xamples/objc/AppRTCDemo/ios/Info.plist
xamples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.h
xamples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.m
xamples/objc/AppRTCDemo/ios/main.m
xamples/objc/AppRTCDemo/ios/resources/Roboto-Regular.ttf
xamples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png
xamples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png
xamples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png
xamples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp.png
xamples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp@2x.png
xamples/objc/AppRTCDemo/ios/resources/ic_clear_black_24dp.png
xamples/objc/AppRTCDemo/ios/resources/ic_clear_black_24dp@2x.png
xamples/objc/AppRTCDemo/ios/resources/ic_switch_video_black_24dp.png
xamples/objc/AppRTCDemo/ios/resources/ic_switch_video_black_24dp@2x.png
xamples/objc/AppRTCDemo/mac/APPRTCAppDelegate.h
xamples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m
xamples/objc/AppRTCDemo/mac/APPRTCViewController.h
xamples/objc/AppRTCDemo/mac/APPRTCViewController.m
xamples/objc/AppRTCDemo/mac/Info.plist
xamples/objc/AppRTCDemo/mac/main.m
xamples/objc/AppRTCDemo/tests/ARDAppClientTest.mm
xamples/objc/AppRTCDemo/third_party/SocketRocket/LICENSE
xamples/objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.h
xamples/objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.m
xamples/objc/Icon.png
xamples/objc/README
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
xamples/peerconnection/client/defaults.cc
xamples/peerconnection/client/defaults.h
xamples/peerconnection/client/flagdefs.h
xamples/peerconnection/client/linux/main.cc
xamples/peerconnection/client/linux/main_wnd.cc
xamples/peerconnection/client/linux/main_wnd.h
xamples/peerconnection/client/main.cc
xamples/peerconnection/client/main_wnd.cc
xamples/peerconnection/client/main_wnd.h
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/client/peer_connection_client.h
xamples/peerconnection/server/data_socket.cc
xamples/peerconnection/server/data_socket.h
xamples/peerconnection/server/main.cc
xamples/peerconnection/server/peer_channel.cc
xamples/peerconnection/server/peer_channel.h
xamples/peerconnection/server/server_test.html
xamples/peerconnection/server/utils.cc
xamples/peerconnection/server/utils.h
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| 5b4ce3391d60229342963ef524b7c1e359e5bfc4 |
05-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
DtlsIdentityStoreInterface added.
New PeerConnectionFactoryInterface::CreatePeerConnection taking both service and store added (old CreatePC signature still exists).
This is CL is part of an effort to land https://codereview.webrtc.org/1176383004 without breaking Chromium.
See bug for more information.
BUG=webrtc:4899
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1268363002 .
Cr-Commit-Position: refs/heads/master@{#9680}
pp/webrtc/dtlsidentitystore.h
pp/webrtc/peerconnectioninterface.h
|
| 0c0226408dc6f42abc2cd53cab2de02d3ee610d7 |
05-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it.
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1270333002 .
Cr-Commit-Position: refs/heads/master@{#9679}
pp/webrtc/statscollector_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
|
| bd10ee8bd3292a33d5218ff2e5103af21f116c63 |
05-Aug-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Tiny cleanups.
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1272163002 .
Cr-Commit-Position: refs/heads/master@{#9678}
edia/base/videoengine_unittest.h
edia/webrtc/webrtcmediaengine.h
|
| 37ec7330b4eff1da0f756fc9ac966d25cbc0fce7 |
05-Aug-2015 |
magjed <magjed@webrtc.org> |
VideoCapturerAndroid: Check if data is null in onPreviewFrame()
onPreviewFrame() might be called with a null data pointer, which is allowed according to the documentation.
BUG=webrtc:4877
Review URL: https://codereview.webrtc.org/1260183004
Cr-Commit-Position: refs/heads/master@{#9674}
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 0c850202fe76ae55401028217302a11a7c0d0a19 |
04-Aug-2015 |
Alex Glaznev <glaznev@google.com> |
Add list of devices with HW H.264 encoder non suitable for WebRTC.
For now add only Galaxy S4 to the list, since its H.264 HW encoder
generates two times lower bitrate comparing to target.
Also use VBR mode for H.264 encoder configuration.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1270603007 .
Cr-Commit-Position: refs/heads/master@{#9673}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 503726c3498201822079c5abe9e528498846c9f2 |
31-Jul-2015 |
honghaiz <honghaiz@webrtc.org> |
Fix the generation mismatch assertion error.
BUG=4860
Review URL: https://codereview.webrtc.org/1248063002
Cr-Commit-Position: refs/heads/master@{#9667}
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| b28678ce70d9e9f57ef50364f2054c4a1d630149 |
26-Jul-2015 |
magjed <magjed@webrtc.org> |
Add unittest to GlRectDrawer
Review URL: https://codereview.webrtc.org/1250093003
Cr-Commit-Position: refs/heads/master@{#9638}
pp/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
pp/webrtc/java/android/org/webrtc/EglBase.java
|
| 013a5800641019c87dac68be4fbc4ab07c1109c1 |
26-Jul-2015 |
magjed <magjed@webrtc.org> |
VideoCapturerAndroid: Revert elapsedRealtimeNanos to elapsedRealtime
Review URL: https://codereview.webrtc.org/1254143002
Cr-Commit-Position: refs/heads/master@{#9637}
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| e2b34b7b4b540c4d92397c20990b7c5dbf64bbc3 |
24-Jul-2015 |
jackychen <jackychen@webrtc.org> |
Bug fix: camera frames are dropped before wideo encoder.
https://code.google.com/p/webrtc/issues/detail?id=4871
R=glaznev@webrtc.org
TBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1260543002 .
Cr-Commit-Position: refs/heads/master@{#9634}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 6bb1b6e7fe5631e9f218b80292df5b64623c5616 |
24-Jul-2015 |
pbos <pbos@webrtc.org> |
Control combined_audio_video_bwe with config bool.
Permits setting RTP extensions for AudioReceiveStream without enabling
combined A/V BWE. This prevents spamming the log with "Failed to find
extension id:".
BUG=webrtc:4870
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1256803004
Cr-Commit-Position: refs/heads/master@{#9633}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| c3f46a9f7fe4536877e7b5f6af109dbdc08b8242 |
23-Jul-2015 |
tkchin <tkchin@webrtc.org> |
iOS: Move AppRTC logging methods to public headers.
BUG=
Review URL: https://codereview.webrtc.org/1241283004
Cr-Commit-Position: refs/heads/master@{#9629}
pp/webrtc/objc/RTCLogging.mm
pp/webrtc/objc/public/RTCLogging.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/ARDSDPUtils.m
xamples/objc/AppRTCDemo/ARDSignalingMessage.m
xamples/objc/AppRTCDemo/ARDWebSocketChannel.m
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.m
xamples/objc/AppRTCDemo/common/ARDLogging.h
xamples/objc/AppRTCDemo/common/ARDLogging.mm
xamples/objc/AppRTCDemo/common/ARDUtilities.m
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
ibjingle.gyp
ibjingle_examples.gyp
|
| 28bae02bd383abbd35cc87fa1188a569f1b3c683 |
23-Jul-2015 |
tkchin <tkchin@webrtc.org> |
Remove CircularFileStream / replace it with CallSessionFileRotatingStream.
BUG=4838, 4839
Review URL: https://codereview.webrtc.org/1245143005
Cr-Commit-Position: refs/heads/master@{#9628}
pp/webrtc/objc/RTCFileLogger.mm
pp/webrtc/objc/public/RTCFileLogger.h
|
| 86c6d33aec684d08189d498912e47cbc17c4d2db |
23-Jul-2015 |
Michael Graczyk <mgraczyk@chromium.org> |
Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
An earlier version of this commit caused a crash on AEC dump.
TBR=aluebs@webrtc.org,pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1248393003 .
Cr-Commit-Position: refs/heads/master@{#9626}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 66f438f8c37165db669e15314cdb7bbfa0841f4c |
23-Jul-2015 |
magjed <magjed@webrtc.org> |
Revert of Fixing scenario where track is rejected and later un-rejected. (patchset #5 id:80001 of https://codereview.webrtc.org/1231613002/)
Reason for revert:
I think this causes WebRtcBrowserTest.CallAndModifyStream to fail on Android. See https://code.google.com/p/webrtc/issues/detail?id=4857 for more info.
Original issue's description:
> Fixing scenario where track is rejected and later un-rejected.
>
> Added `RestartLocalTracks` and `RestartRemoteTracks` methods to
> `MediaStreamHandlerContainer` which will redo the track handlers'
> initial setup; most importantly, this will re-connect the
> renderer/capturer/etc. to a channel which was destroyed and then
> re-created.
>
> Also added `AcceptRemoteTracks` method to MediaStreamSignaling, which
> does the inverse of `RejectRemoteTracks`. Effectively this will notify
> sinks that the track is live again, after previously being set to
> `kEnded` when it was rejected.
>
> BUG=webrtc:2136
>
> Committed: https://crrev.com/be37888b6d5d269dbd5385569dba15c0d70594f2
> Cr-Commit-Position: refs/heads/master@{#9600}
TBR=pthatcher@webrtc.org,juberti@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1247443005
Cr-Commit-Position: refs/heads/master@{#9622}
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/peerconnection.cc
|
| 64e753c3998a17429418180b3a947231a9fd98cd |
23-Jul-2015 |
magjed <magjed@webrtc.org> |
Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 id:240001 of https://codereview.webrtc.org/1226093007/)
Reason for revert:
Breaks Chromium FYI content_browsertest on all platforms. The testcase that fails is WebRtcAecDumpBrowserTest.CallWithAecDump.
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/19388
Sample output:
[ RUN ] WebRtcAecDumpBrowserTest.CallWithAecDump
Xlib: extension "RANDR" missing on display ":9".
[4:14:0722/211548:1282124453:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: ISAC/48000/1 (105)
[4:14:0722/211548:1282124593:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMU/8000/2 (110)
[4:14:0722/211548:1282124700:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: PCMA/8000/2 (118)
[4:14:0722/211548:1282124815:WARNING:webrtcvoiceengine.cc(472)] Unexpected codec: G722/8000/2 (119)
[19745:19745:0722/211548:1282133667:INFO:CONSOLE(64)] "Looking at video in element remote-view-1", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
[19745:19745:0722/211548:1282136892:INFO:CONSOLE(64)] "Looking at video in element remote-view-2", source: http://127.0.0.1:48819/media/webrtc_test_utilities.js (64)
../../content/test/webrtc_content_browsertest_base.cc:62: Failure
Value of: ExecuteScriptAndExtractString( shell()->web_contents(), javascript, &result)
Actual: false
Expected: true
Failed to execute javascript call({video: true, audio: true});.
From javascript: (nothing)
When executing 'call({video: true, audio: true});'
../../content/test/webrtc_content_browsertest_base.cc:75: Failure
Failed
../../content/browser/media/webrtc_aecdump_browsertest.cc:26: Failure
Expected: (base::kNullProcessId) != (*id), actual: 0 vs 0
../../content/browser/media/webrtc_aecdump_browsertest.cc:95: Failure
Value of: GetRenderProcessHostId(&render_process_id)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:99: Failure
Value of: base::PathExists(dump_file)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:101: Failure
Value of: base::GetFileSize(dump_file, &file_size)
Actual: false
Expected: true
../../content/browser/media/webrtc_aecdump_browsertest.cc:102: Failure
Expected: (file_size) > (0), actual: 0 vs 0
[ FAILED ] WebRtcAecDumpBrowserTest.CallWithAecDump, where TypeParam = and GetParam() = (361 ms)
Original issue's description:
> Allow more than 2 input channels in AudioProcessing.
>
> The number of output channels is constrained to be equal to either 1 or the
> number of input channels.
>
> R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/c204754b7a0cc801c70e8ce6c689f57f6ce00b3b
TBR=andrew@webrtc.org,aluebs@webrtc.org,ajm@chromium.org,pbos@chromium.org,pbos@webrtc.org,mgraczyk@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1253573005
Cr-Commit-Position: refs/heads/master@{#9621}
edia/webrtc/fakewebrtcvoiceengine.h
|
| c204754b7a0cc801c70e8ce6c689f57f6ce00b3b |
23-Jul-2015 |
Michael Graczyk <mgraczyk@chromium.org> |
Allow more than 2 input channels in AudioProcessing.
The number of output channels is constrained to be equal to either 1 or the
number of input channels.
R=aluebs@webrtc.org, andrew@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1226093007 .
Cr-Commit-Position: refs/heads/master@{#9619}
edia/webrtc/fakewebrtcvoiceengine.h
|
| b69ab79338bff71ea411b82f3dd59508617a11d5 |
22-Jul-2015 |
magjed <magjed@webrtc.org> |
VideoCapturerAndroid: Add function to change capture format while camera is running
Review URL: https://codereview.webrtc.org/1178703009
Cr-Commit-Position: refs/heads/master@{#9608}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| be37888b6d5d269dbd5385569dba15c0d70594f2 |
17-Jul-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing scenario where track is rejected and later un-rejected.
Added `RestartLocalTracks` and `RestartRemoteTracks` methods to
`MediaStreamHandlerContainer` which will redo the track handlers'
initial setup; most importantly, this will re-connect the
renderer/capturer/etc. to a channel which was destroyed and then
re-created.
Also added `AcceptRemoteTracks` method to MediaStreamSignaling, which
does the inverse of `RejectRemoteTracks`. Effectively this will notify
sinks that the track is live again, after previously being set to
`kEnded` when it was rejected.
BUG=webrtc:2136
Review URL: https://codereview.webrtc.org/1231613002
Cr-Commit-Position: refs/heads/master@{#9600}
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/peerconnection.cc
|
| fabe2c961f9cf86d519532a96e96fa7d6c4ca37d |
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Remove deprecated functions.
This CL removes some functions that are marked as deprecated. Chromium
has been updated in https://crrev.com/7dee3f68b7699ad72c7fc4d75332f72703313849
to call the new functions.
Review URL: https://codereview.webrtc.org/1237613003
Cr-Commit-Position: refs/heads/master@{#9598}
pp/webrtc/jsep.h
pp/webrtc/jsepicecandidate.cc
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/objc/RTCICECandidate.mm
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
xamples/peerconnection/client/conductor.cc
edia/base/streamparams.h
|
| c27d89fdc6b33846ff06e8ca8bd03119d05c6530 |
16-Jul-2015 |
qiangchen <qiangchen@chromium.org> |
Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.
Review URL: https://codereview.webrtc.org/1225153002
Cr-Commit-Position: refs/heads/master@{#9597}
edia/base/fakevideocapturer.h
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| bd3842808996dbb85007242214352f1e6ebd3d17 |
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Don't use result of "field_trial::FindFullName" as string reference.
"field_trial::FindFullName" can return "std::string()" which should not
be referenced by the caller.
Review URL: https://codereview.webrtc.org/1238943003
Cr-Commit-Position: refs/heads/master@{#9594}
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine2.cc
|
| a9b4c32052fd55df7e1d02e846fbea3178bebf71 |
16-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1226093010 .
Cr-Commit-Position: refs/heads/master@{#9593}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine2_unittest.cc
ession/media/channel.cc
ession/media/channel_unittest.cc
|
| 083b73fb95755b78cb0b9cbe67752b7e7b7eb263 |
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Use std::string references instead of copying contents.
This CL improves the memory footprint a bit by using string references
instead of creating a copy.
Review URL: https://codereview.webrtc.org/1241973002
Cr-Commit-Position: refs/heads/master@{#9592}
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/webrtcsdp.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine2.cc
ession/media/mediasession.cc
|
| cd6702282a49448adda470934f4bd9e6181cab22 |
16-Jul-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
Define Stream base classes
BUG=webrtc:4690
Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1226123005 .
Cr-Commit-Position: refs/heads/master@{#9591}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| f39382943449b7e44ac563e05a14203534591acf |
15-Jul-2015 |
deadbeef <deadbeef@webrtc.org> |
Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used.
Tested that this doesn't break compatibility with Firefox or older
versions of Chrome, no matter which side generates the initial offer.
BUG=webrtc:2796
Review URL: https://codereview.webrtc.org/1219333002
Cr-Commit-Position: refs/heads/master@{#9589}
pp/webrtc/webrtcsession_unittest.cc
ession/media/mediasession.cc
|
| 8fc7fa798f7a36955f1b933980401afad2aff592 |
15-Jul-2015 |
pbos <pbos@webrtc.org> |
Base A/V synchronization on sync_labels.
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.
BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1181653002
Cr-Commit-Position: refs/heads/master@{#9586}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 2d3b7e2173c672dca5d97d9a5c8ab4217652c442 |
14-Jul-2015 |
Zeke Chin <tkchin@webrtc.org> |
AppRTCDemo file logging.
Adds logging macros to log logs to a file. Undeletes CircularFileStream
for that purpose.
BUG=
R=jiayl@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1217473011 .
Cr-Commit-Position: refs/heads/master@{#9582}
pp/webrtc/objc/RTCFileLogger.mm
pp/webrtc/objc/public/RTCFileLogger.h
xamples/objc/.clang-format
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/ARDSDPUtils.m
xamples/objc/AppRTCDemo/ARDSignalingMessage.m
xamples/objc/AppRTCDemo/ARDUtilities.h
xamples/objc/AppRTCDemo/ARDUtilities.m
xamples/objc/AppRTCDemo/ARDWebSocketChannel.m
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.m
xamples/objc/AppRTCDemo/common/ARDLogging.h
xamples/objc/AppRTCDemo/common/ARDLogging.mm
xamples/objc/AppRTCDemo/common/ARDUtilities.h
xamples/objc/AppRTCDemo/common/ARDUtilities.m
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
ibjingle.gyp
ibjingle_examples.gyp
|
| a03cd3fdef6c823e89c76bb097abd8b83285e4da |
14-Jul-2015 |
honghaiz <honghaiz@webrtc.org> |
1. Override and virtual has to be consistent.
2. provide an implementation for SetIceConnectionReceivingTimeout so that Chrome does not complain.
BUG=
Review URL: https://codereview.webrtc.org/1227843006
Cr-Commit-Position: refs/heads/master@{#9574}
pp/webrtc/peerconnectioninterface.h
|
| 6e2ce6e1ae41d8eeb0f233cbd26087daa03ab702 |
14-Jul-2015 |
jackychen <jackychen@webrtc.org> |
Allow for framerate reduction for HW encoder.
R=pbos@webrtc.org, stefan@webrtc.org
TBR=glaznev@google.com
Review URL: https://webrtc-codereview.appspot.com/51159004 .
Cr-Commit-Position: refs/heads/master@{#9573}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 900996290c996193ac3e418f315354fd2bd0ea8a |
13-Jul-2015 |
honghaiz <honghaiz@webrtc.org> |
Add methods to set the ICE connection receiving_timeout values.
BUG=
Review URL: https://codereview.webrtc.org/1231913003
Cr-Commit-Position: refs/heads/master@{#9572}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
|
| d10a68e7974a29b26d6c926e6f137255f3c173be |
10-Jul-2015 |
noahric <noahric@chromium.org> |
Don't create unsignalled receive streams for RTX, RED RTX, and ULPFEC packets.
BUG=webrtc:4389
Review URL: https://codereview.webrtc.org/1226093002
Cr-Commit-Position: refs/heads/master@{#9566}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| a6d2444c84004d10a5d8b8517bbd178600f8412f |
10-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1228203002 .
Cr-Commit-Position: refs/heads/master@{#9564}
pp/webrtc/webrtcsession.h
ession/media/channel.cc
ession/media/channel_unittest.cc
|
| bb36fdf95f9667fb1f3fbf3073bd15007681322c |
09-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove empty-string comparisons.
Use .empty() and !.empty() in favor of == "" or != "".
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1228913003
Cr-Commit-Position: refs/heads/master@{#9559}
pp/webrtc/webrtcsdp.cc
|
| 3b1e647b6a6f74d8e4392e012fe2262b3d2c4334 |
09-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove media sinks from Channel.
Allows removing MediaRecorder which isn't in use apart from channel
unittests, along with it unittests for MediaRecorder that are flaky when
run in parallel can also go.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1219663008
Cr-Commit-Position: refs/heads/master@{#9558}
ibjingle.gyp
ibjingle_tests.gyp
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediarecorder.cc
ession/media/mediarecorder.h
ession/media/mediarecorder_unittest.cc
|
| 0f620f4e318a162e7a251133e7a8ddea5188b9bb |
09-Jul-2015 |
tommi <tommi@webrtc.org> |
Make sure we process all pending offer/answer requests before terminating.
This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.
BUG=chromium:507307
Review URL: https://codereview.webrtc.org/1231823002
Cr-Commit-Position: refs/heads/master@{#9557}
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| 61093868b4b37f5b3212d5d682c957357d568026 |
09-Jul-2015 |
Jiayang Liu <jiayl@chromium.org> |
Expose the disable encryption option to JNI.
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1230613002 .
Cr-Commit-Position: refs/heads/master@{#9554}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| 54360510ff9b7c61fc906d3ed360b06a5824bbf1 |
08-Jul-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add flakyness check based on the recently received packets.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1207563002 .
Cr-Commit-Position: refs/heads/master@{#9553}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 4e7aa43ea0fd7106cd39036798877301398966a6 |
07-Jul-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
audio_processing: Adds two UMA histograms logging delay jumps in AEC
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.
This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.
Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.
BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229443003.
Cr-Commit-Position: refs/heads/master@{#9544}
edia/webrtc/fakewebrtcvoiceengine.h
|
| ac8869ec5a606e0a0ab71e70937c8fbf403630ce |
03-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Report metrics about negotiated ciphers.
This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).
BUG=428343
Review URL: https://codereview.webrtc.org/1156143005
Cr-Commit-Position: refs/heads/master@{#9537}
pp/webrtc/fakemetricsobserver.cc
pp/webrtc/fakemetricsobserver.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ibjingle_tests.gyp
|
| 0f133b99c655cbdb347b4a71ac872c071532189f |
02-Jul-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Rename APM Config ReportedDelay to DelayAgnostic
We use this Config struct for enabling/disabling the delay agnostic
AEC. This change renames it to DelayAgnostic for readability reasons.
NOTE: The logic is reversed in this CL. The old ReportedDelay config
turned DA-AEC off, while the new DelayAgnostic turns it on.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, ReportedDelay is disabled or DelayAgnostic is enabled, DA-AEC
is engaged in APM.
BUG=webrtc:4651
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1211053006
Cr-Commit-Position: refs/heads/master@{#9531}
edia/webrtc/webrtcvoiceengine.cc
|
| 0d7dbde8cf57b0492c6988cb3bdca9933ce86d55 |
02-Jul-2015 |
tkchin <tkchin@webrtc.org> |
Update AppRTCDemo resolution for iPhone6/6+
BUG=
Review URL: https://codereview.webrtc.org/1214113015
Cr-Commit-Position: refs/heads/master@{#9530}
xamples/objc/AppRTCDemo/ios/Info.plist
xamples/objc/AppRTCDemo/ios/resources/Default-568h.png
xamples/objc/AppRTCDemo/ios/resources/iPhone5@2x.png
xamples/objc/AppRTCDemo/ios/resources/iPhone6@2x.png
xamples/objc/AppRTCDemo/ios/resources/iPhone6p@3x.png
ibjingle_examples.gyp
|
| 0edd50ccb34cc2dc4746137fdce1f5cf66808274 |
01-Jul-2015 |
bemasc <bemasc@webrtc.org> |
Support for onbufferedamountlow
Original review at https://webrtc-codereview.appspot.com/54679004/
BUG=https://code.google.com/p/chromium/issues/detail?id=496700
Review URL: https://codereview.webrtc.org/1207613006
Cr-Commit-Position: refs/heads/master@{#9527}
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/DataChannel.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/objc/public/RTCDataChannel.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 71f6f4405c1c5f60097f8d10841378088e78e8b9 |
29-Jun-2015 |
Zeke Chin <tkchin@webrtc.org> |
iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.
BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1187573004.
Cr-Commit-Position: refs/heads/master@{#9515}
pp/webrtc/objc/RTCPeerConnectionFactory.mm
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDSDPUtils.h
xamples/objc/AppRTCDemo/ARDSDPUtils.m
xamples/objc/AppRTCDemo/tests/ARDAppClientTest.mm
ibjingle_examples.gyp
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvideoengine2.cc
|
| 4b91bd08979fcfb191cdae27ad24936beefce735 |
26-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Move frame input (ViECapturer) to webrtc/video/.
Renames ViECapturer to VideoCaptureInput and initializes several
parameters on construction instead of setters.
Also removes an old deadlock suppression.
BUG=1695, 2999
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53559004.
Cr-Commit-Position: refs/heads/master@{#9508}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvideoengine.h
|
| c0c3a865f49a6386b0815001b9856c1eee27e7c2 |
25-Jun-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Prevent JS from bypassing RTP data channel bandwidth limitation.
Normally the RTP data channel is capped at 30kbps, but by mangling the
SDP string, one could get around this limitation. With this fix,
SdpDeserialize will return an error if it detects this condition.
BUG=280726
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1196403004.
Cr-Commit-Position: refs/heads/master@{#9499}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 8d3e489d01654a4107b99fcfb4ee9ffff9d24d38 |
24-Jun-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Update deeper codereview.settings files to match the root.
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1190883002.
Cr-Commit-Position: refs/heads/master@{#9498}
odereview.settings
|
| 59a677ada27c660e9cd7486f0d702753dbeb6d39 |
24-Jun-2015 |
magjed <magjed@webrtc.org> |
Android VideoRendererGui: Refactor GLES rendering
This CL should not change any visible behaviour. It does the following:
* Extract GLES rendering into separate class GlRectDrawer. This class is also needed for future video encode with OES texture input.
* Clean up current ScalingType -> display size calculation and introduce new SCALE_ASPECT_BALANCED (b/21735609) and remove unused SCALE_FILL.
* Replace current mirror/rotation index juggling with android.opengl.Matrix operations instead.
Review URL: https://codereview.webrtc.org/1191243005
Cr-Commit-Position: refs/heads/master@{#9496}
pp/webrtc/java/android/org/webrtc/GlRectDrawer.java
pp/webrtc/java/android/org/webrtc/GlShader.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
ibjingle.gyp
|
| 2c4c9148191a10c0e82c9a209d454c6b1ebbaf20 |
24-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
In screenshare mode, suppress VP8 bitrate overshoot and increase quality
This change includes several improvements:
* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers
A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.
BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1193513006.
Cr-Commit-Position: refs/heads/master@{#9495}
edia/webrtc/simulcast.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 7ab5f801dd8d6bc018b59d41877f44ec4ab19d15 |
24-Jun-2015 |
phoglund <phoglund@webrtc.org> |
Adding an equals method for KeyValuePair for easier testing.
With this we can write stuff like
assertThat(result.mandatory,
hasItem(new KeyValuePair("minWidth", "1280")));
The above will currently fail because the object falls back to ==.
BUG=None
Review URL: https://codereview.webrtc.org/1193883006
Cr-Commit-Position: refs/heads/master@{#9494}
pp/webrtc/java/src/org/webrtc/MediaConstraints.java
|
| 66f920ea57c76e6213ada45ad907872f4fa2e7ee |
24-Jun-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Remove definition of non-existent method.
The private method "CreateDefaultLocalDescription" is defined in the
class, but not implemented or used anywhere.
R=juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1182793004.
Cr-Commit-Position: refs/heads/master@{#9493}
pp/webrtc/webrtcsession.h
|
| 39b31001d2d74373f1e60cdbc8e57e402bdf34e3 |
23-Jun-2015 |
tommi <tommi@webrtc.org> |
Change kEchoCancellation to be 'echoCancellation'.
This is the second cl in WebRTC for this change and will be landed after Chromium has been updated to use kGooglEchoCancellation where that variant is required. See also the first part: https://codereview.webrtc.org/1179233003
BUG=webrtc:4747
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1185963003
Cr-Commit-Position: refs/heads/master@{#9490}
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
|
| be24c94c95056e4f0a22039f25f2fa8a27be6b66 |
23-Jun-2015 |
jbauch <jbauch@webrtc.org> |
Set / verify stats report timestamps.
This CL updates the track report timestamps which were fixed at "0" before
and updates the timestamps in reports for local audio tracks.
Also the timestamps are checked in various tests to make sure no "0" is
returned.
Original CL is at https://webrtc-codereview.appspot.com/51829004/
BUG=webrtc:4316
TBR=hta@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1204493002
Cr-Commit-Position: refs/heads/master@{#9485}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 1d34fe979c52e5826c5c8162759b0167b2607836 |
16-Jun-2015 |
henrika <henrika@chromium.org> |
Adds support for webrtc::test::ResourcePath on iOS
BUG=webrtc:4752
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1178843002.
Cr-Commit-Position: refs/heads/master@{#9445}
pp/webrtc/objc/RTCDataChannel.mm
|
| b02af18c5cb6d6c3def7f44d27a63068360f4f29 |
16-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Follow-up: Remove old DelayCorrection AEC config
This is a follow-up to r9401, where the configuration DelayCorrection
was replaced by ExtendedFilter.
This change also removes the media constraint
kExperimentalEchoCancellation which was replaced by
kExtendedFilterEchoCancellation in the same CL.
Both settings that are now being removed were kept in the code to avoid
API breakages. In https://codereview.chromium.org/1167343004,
depending code has been updated to avoid breakages.
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1181413004.
Cr-Commit-Position: refs/heads/master@{#9444}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
|
| 05ce5dd0f159c43b50bd87f7faf1fbe8f31d5845 |
15-Jun-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Roll chromium_revision e937e5f..c2239a8 (333350:334133)
Removed no longer used test_isolation_outdir variable as in
https://codereview.chromium.org/1176463003
The move of a DEPS in https://codereview.chromium.org/1155743013
is causing problems on some trybots. It shouldn't affect developers.
Relevant changes:
* src/third_party/android_tools: a3afc68..ed3dde6
* src/third_party/icu: 9939a5d..a05f412
* src/third_party/libjpeg_turbo: 8ee9bdd..f4631b6
* src/third_party/libyuv: 632c50f..632c50f
Details: https://chromium.googlesource.com/chromium/src/+/e937e5f..c2239a8/DEPS
Clang version was not updated in this roll.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1182043002.
Cr-Commit-Position: refs/heads/master@{#9435}
uild/isolate.gypi
|
| 2b679250fbd50b3c8d9ac266a42fbc8a1bd84167 |
15-Jun-2015 |
Åsa Persson <asapersson@webrtc.org> |
VideoCapturerAndroid: Add possibility to request a new resolution from the video adapter.
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1178643006.
Cr-Commit-Position: refs/heads/master@{#9434}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 70c7fe14ac322a3abac2726d29ae3264b956e3da |
15-Jun-2015 |
Tommi <tommi@webrtc.org> |
Add kGoogEchoCancellation to MediaConstraintsInterface.
This constraint will be equal to kEchoCancellation until we've updated Chromium to use kGoogEchoCancellation where that constraint is needed. Once that's done, I'll change kEchoCancellation to be 'echoCancellation'.
BUG=webrtc:4747
R=andrew@webrtc.org
Review URL: https://codereview.webrtc.org/1179233003.
Cr-Commit-Position: refs/heads/master@{#9433}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
|
| 782671f7983e27be865f04f546fbfc3f1cb4e1b3 |
13-Jun-2015 |
Alex Glaznev <glaznev@google.com> |
Improve Android HW decoder error handling.
- Remove an option to use MediaCodec SW decoder from Java layer.
- Better handling Java exceptions in JNI - detect exceptions
and either try to reset the codec or fallback to SW decoder.
- If any error is reported by codec try to fallback to SW
codec for VP8 or reset decoder and continue decoding for H.264.
- Add more logging for error conditions.
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1178943007.
Cr-Commit-Position: refs/heads/master@{#9431}
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 2f65ac143782ee7f57b2b5ee3a9981d5843ed1f8 |
12-Jun-2015 |
Jon Hjelle <hjon@andyet.net> |
Fix crash and warning in AppRTCDemo
Don't dismiss the presented view controller if it's already being dismissed to clear a warning about dismissing from a view controller while a dismiss is in progress.
Remove the sample buffer delegate when capture is being stopped to avoid a crash when a delegate method is sent to a deallocated object.
BUG=webrtc:4734
R=jiayl@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54669004.
Patch from Jon Hjelle <hjon@andyet.net>.
Cr-Commit-Position: refs/heads/master@{#9430}
pp/webrtc/objc/avfoundationvideocapturer.mm
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
|
| 728d9037c016c01295177fa700fc7927f0bb80bb |
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
|
| b7e5054414ff524f9db81dab7917729b8c4c8bcb |
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
|
| eb82309d066305e5d8e94ba1f58a4cbdbcb4f9bb |
11-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Remove FileMediaEngine.
This is currently only used in libjingle/examples/call which is
deprecated and not currently building.
BUG=
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1169833004.
Cr-Commit-Position: refs/heads/master@{#9415}
ibjingle.gyp
ibjingle_tests.gyp
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
|
| 80cf97cddd9c67fddb8cd9f78e7d560a2c0deec0 |
11-Jun-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Android rendering: Move common EGL and GL functions to separate classes
This CL does not make any functional changes. The purpose is to extract some common code that is needed for texture capture and texture encode.
This CL does the following changes:
* Move common EGL functions from org.webrtc.MediaCodecVideoDecoder to org.webrtc.EglBase.
* Move common GL functions from org.webrtc.VideoRendererGui to org.webrtc.GlUtil and org.webrtc.GlShader.
* Remove unused call to surfaceTexture.getTransformMatrix in YuvImageRenderer.
* Add helper functions rotatedWidth()/rotatedHeight() in VideoRenderer.I420Frame.
R=glaznev@webrtc.org, hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47309005.
Cr-Commit-Position: refs/heads/master@{#9414}
pp/webrtc/java/android/org/webrtc/EglBase.java
pp/webrtc/java/android/org/webrtc/GlShader.java
pp/webrtc/java/android/org/webrtc/GlUtil.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
ibjingle.gyp
|
| f045e4da43e671ae511aa1d9b6ef2968256a745d |
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Prepare to convert various types to size_t.
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question. This is preparation for a future change
that will convert a variety of types to size_t.
There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.
BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm
Review URL: https://codereview.webrtc.org/1174813003
Cr-Commit-Position: refs/heads/master@{#9413}
pp/webrtc/test/fakeaudiocapturemodule.h
|
| 8a19f3dc62f4817404c322280d8f035de1adb56a |
10-Jun-2015 |
Niklas Enbom <niklas.enbom@webrtc.org> |
Relanding https://webrtc-codereview.appspot.com/56589004
BUG=
TBR=cpaulin@chromium.org
Review URL: https://codereview.webrtc.org/1176023002.
Cr-Commit-Position: refs/heads/master@{#9410}
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 54b0ca553f891acbfd44ec1f659663f618df4e80 |
10-Jun-2015 |
Niklas Enbom <niklas.enbom@webrtc.org> |
Revert "Landing https://webrtc-codereview.appspot.com/53669004/"
This reverts commit 2aef19cbde01cb975eb3d6100610d31bbbae9258.
BUG=
TBR=cpaulin@chromium.org
Review URL: https://codereview.webrtc.org/1168313003.
Cr-Commit-Position: refs/heads/master@{#9404}
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 2aef19cbde01cb975eb3d6100610d31bbbae9258 |
10-Jun-2015 |
Niklas Enbom <niklas.enbom@webrtc.org> |
Landing https://webrtc-codereview.appspot.com/53669004/
BUG=
Review URL: https://codereview.webrtc.org/1169123003.
Cr-Commit-Position: refs/heads/master@{#9403}
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 532caeae2d3e8c043f61f72d2f909c081ce2e6b4 |
09-Jun-2015 |
Tommi <tommi@webrtc.org> |
Adding DCHECKs and constness to DtlsIdentityStore.
R=hbos@webrtc.org, hbos
BUG=
Review URL: https://codereview.webrtc.org/1171893003.
Cr-Commit-Position: refs/heads/master@{#9402}
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
|
| 441f6347311bcf2079435c3888d67e1fb321f9f8 |
09-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)
The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.
Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec."
BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1151573021.
Cr-Commit-Position: refs/heads/master@{#9401}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 04f4931ef06273c2873e7816ed1f568d445117b8 |
08-Jun-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
VoE2 API draft
BUG=4690
R=jmarusic@webrtc.org, kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50029004
Cr-Commit-Position: refs/heads/master@{#9392}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
|
| d7da120b40f7a8a8357f23cf6b49aa03f67c1cf6 |
05-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Disable reduced-size RTCP in default config.
Verifies that reduced-size isn't configured in WebRtcVideoEngine2
without explicit configuration (which doesn't exist). Also disables REMB
in the default config because it requires reconfiguration.
Adds default-config tests to make sure that they don't contain
parameters that need to be negotiated between clients.
BUG=chromium:497103, webrtc:4745
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1171533002
Cr-Commit-Position: refs/heads/master@{#9384}
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| fe55c38effd2984913aa4d0f53a12f46b9a56fd4 |
05-Jun-2015 |
henrika <henrika@chromium.org> |
Removes automatic setting of COMM mode in WebRTC.
It is now up to the application to ensure that it is in COMM mode before any audio streaming is started.
BUG=b/21571563
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1165923002
Cr-Commit-Position: refs/heads/master@{#9383}
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
|
| eb66e800d1f5f74ab366715d2618fbede8cf3e12 |
05-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Re-land "Convert native handles to buffers before encoding."
This reverts commit a67675506c9057bd9ffd4d76aae8b743343d434d.
BUG=webrtc:4081
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1158273010
Cr-Commit-Position: refs/heads/master@{#9381}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 3fbf3f8841b5460503fb646eaedcb063620434a8 |
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.
BUG=4696
TBR=bjornv@webrtc.org
Review URL: https://codereview.webrtc.org/1166463006
Cr-Commit-Position: refs/heads/master@{#9380}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16 |
05-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.
The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.
This change also renames experimental_aec in AudioOptions to extended_filter_aec.
BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54659004
Cr-Commit-Position: refs/heads/master@{#9378}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| a9952cdd0e8d909219f0463caed11265da028ced |
03-Jun-2015 |
Tommi <tommi@webrtc.org> |
Remove CHECK from GetThreadName.
It's safe for prctl() to fail, so we fall back on <noname> for thread names if we can't get one, instead of crashing.
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57529004
Cr-Commit-Position: refs/heads/master@{#9363}
pp/webrtc/java/jni/jni_helpers.cc
|
| 73f72105c4b671624613cc132bfa86cfc956318b |
03-Jun-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
Actively turns off platform-AEC when DA-AEC is used
When initiating a call default audio options are applied, which turns on platform-AEC if such exists. Then, if delay agnostic AEC (DA-AEC) is enabled through a media constraint no action with respect to platform-AEC is taken (a bug) and turning on SW AEC. Hence, we run both AECs.
This CL makes sure the platform-AEC is disabled if we want to run DA-AEC.
BUG=
TESTED=locally on Nexus 4 and Nexus 6.
R=henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52049004
Cr-Commit-Position: refs/heads/master@{#9361}
edia/webrtc/webrtcvoiceengine.cc
|
| 6b990744d9c3b4aa4ce5c4db585acff7285c75cc |
02-Jun-2015 |
Wan-Teh Chang <wtc@chromium.org> |
Revert "Import org.junit.Assert instead of junit.framework.Assert."
This reverts commit a88470964c55dc655022d1f46370565aa3be535f.
It broke Android builds:
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46: error: package org.junit does not exist
import static org.junit.Assert.*;
^
TBR=glaznev@webrtc.org,pthatcher@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/52039004
Cr-Commit-Position: refs/heads/master@{#9357}
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
|
| a88470964c55dc655022d1f46370565aa3be535f |
02-Jun-2015 |
Wan-Teh Chang <wtc@chromium.org> |
Import org.junit.Assert instead of junit.framework.Assert.
This fixed the warning:
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46: warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;
R=glaznev@webrtc.org, pthatcher@webrtc.org
BUG=none
Review URL: https://webrtc-codereview.appspot.com/50209004
Cr-Commit-Position: refs/heads/master@{#9356}
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
|
| 308d163c715df7b4348a1e00bf2a6761c0adb689 |
02-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Revert "Convert native handles to buffers before encoding."
This reverts commit a831dc3a7d10a1fbaa258ee6b1ca6cfc7e91c5ca to unblock
rolling into Chromium.
BUG=4081
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55549004
Cr-Commit-Position: refs/heads/master@{#9354}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 8e6fd46cc324f8946e68396edcb252ffaf2d4579 |
02-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50179004
Cr-Commit-Position: refs/heads/master@{#9350}
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| a831dc3a7d10a1fbaa258ee6b1ca6cfc7e91c5ca |
01-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Convert native handles to buffers before encoding.
Required to permit conversion of NV12 handles on iOS to I420 for VP8
software encoding, which blocks texture-based capture. This change
enforces that all texture-based input provides a method for converting
native handles to I420 if they are ever used with software encoders that
do not understand the native handles.
BUG=4081
R=emircan@chromium.org, glaznev@webrtc.org, hbos@webrtc.org, magjed@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50909005
Cr-Commit-Position: refs/heads/master@{#9347}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 5263b3c1ddb10ecca58d9f08364aad2d6ba1d95d |
01-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Add options for NetEq fast accelerate mode through libjingle
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.
When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.
BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55479004
Cr-Commit-Position: refs/heads/master@{#9344}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 4765070b8d6f024509c717c04d9b708750666927 |
30-May-2015 |
Miguel Casas-Sanchez <mcasas@webrtc.org> |
Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.
Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.
BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52629004
Cr-Commit-Position: refs/heads/master@{#9339}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.cc
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcpassthroughrender.h
edia/webrtc/webrtcpassthroughrender_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| c2cb266c93d5918f7a54fd62c07c387dde9186a2 |
30-May-2015 |
Jon Hjelle <hjon@andyet.net> |
Match video orientation with device orientation for portrait and portrait upside down
BUG=
R=tkchin@webrtc.org
Committed: https://crrev.com/14c2695f2968d6e8546545a9b62940563073b4b6
Patch from Jon Hjelle <hjon@andynet.net>.
Cr-Commit-Position: refs/heads/master@{#9336}
Review URL: https://webrtc-codereview.appspot.com/55459004
Patch from Jon Hjelle <hjon@andyet.net>.
Cr-Commit-Position: refs/heads/master@{#9338}
pp/webrtc/objc/avfoundationvideocapturer.mm
|
| 7be99bdea13d8863a26a4b0b30b8415785809334 |
30-May-2015 |
Zeke Chin <tkchin@webrtc.org> |
Revert "Match video orientation with device orientation for portrait and portrait upside down"
Misspelt contributor's email address. Easier to revert and reland.
TBR=hjon@andyet.net
This reverts commit 14c2695f2968d6e8546545a9b62940563073b4b6.
BUG=
Review URL: https://webrtc-codereview.appspot.com/54619004
Cr-Commit-Position: refs/heads/master@{#9337}
pp/webrtc/objc/avfoundationvideocapturer.mm
|
| 14c2695f2968d6e8546545a9b62940563073b4b6 |
30-May-2015 |
Jon Hjelle <hjon@andynet.net> |
Match video orientation with device orientation for portrait and portrait upside down
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55459004
Patch from Jon Hjelle <hjon@andynet.net>.
Cr-Commit-Position: refs/heads/master@{#9336}
pp/webrtc/objc/avfoundationvideocapturer.mm
|
| bc7dd7e02353556a4206849ae72beb9b3f256706 |
29-May-2015 |
Zeke Chin <tkchin@webrtc.org> |
Add RTCConfiguration constructor to RTCPeerConnection wrapper.
BUG=4658
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56419004
Cr-Commit-Position: refs/heads/master@{#9335}
pp/webrtc/objc/.clang-format
pp/webrtc/objc/RTCEnumConverter.h
pp/webrtc/objc/RTCEnumConverter.mm
pp/webrtc/objc/RTCPeerConnection+Internal.h
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCPeerConnectionInterface+Internal.h
pp/webrtc/objc/RTCPeerConnectionInterface.mm
pp/webrtc/objc/public/RTCPeerConnectionFactory.h
pp/webrtc/objc/public/RTCPeerConnectionInterface.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ios/Info.plist
ibjingle.gyp
|
| d935f912b10e91f9c258b6d9d85594ff2ca7186a |
29-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Don't try to parse empty Ice urls.
https://crrev.com/7c4e7458b5ce99c13a75d5be7d718ef94e2f8f9f added support
to pass a list of urls for IceServer configurations. This CL fixes a
potential crash when empty urls are passed.
BUG=2096
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51969004
Cr-Commit-Position: refs/heads/master@{#9334}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory_unittest.cc
|
| a8202aadd5b33597f55cec12381447a4cf043f3e |
29-May-2015 |
Henrik Kjellander <kjellander@google.com> |
Roll chromium_revision 1b9c098..ccef3cb (330302:331232)
Relevant changes:
* src/buildtools: b73e5f7..dc487f4
* src/third_party/android_tools: 3445d55..3c5189b
* src/third_party/boringssl/src: 9660032..a7997f1
Details: https://chromium.googlesource.com/chromium/src/+/1b9c098..ccef3cb/DEPS
Clang version was not updated in this roll.
BUG=4695
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53499004
Cr-Commit-Position: refs/heads/master@{#9333}
ibjingle.gyp
ibjingle_examples.gyp
|
| 5c6c6e026bbcc83ad546d00b41ab739dcc856c1b |
29-May-2015 |
Lally Singh <lally@google.com> |
Implements TODOs for webrtc::datachannel state management when the SCTP association is congested. Adds missing state variables for each step in the transitions between DataChannelInterface::DataStates (kConnecting, kOpen, etc.), and uses them.
BUG=https://code.google.com/p/chromium/issues/detail?id=474650
R=jiayl@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44299004
Cr-Commit-Position: refs/heads/master@{#9331}
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
|
| c28a896a7bbd8a1ffef44a1f66ac67c43b4eeada |
29-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
BUG=4690
Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code
R=solenberg@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56499004
Cr-Commit-Position: refs/heads/master@{#9330}
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/mediaengine.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 04e5b498278c633bc3c49da43d08c15b1e75ebc0 |
29-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Make maximum SSL version configurable through PeerConnectionFactory::Options
This can be used to activate DTLS 1.2 through a command-line flag from Chromium
later.
BUG=chromium:428343
R=jiayl@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/54509004
Cr-Commit-Position: refs/heads/master@{#9328}
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
|
| e70028e43fded41310d41cc93b90f2e689c04725 |
29-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Protect access to shared list of SRTP sessions.
This is a follow up to https://webrtc-codereview.appspot.com/47319004/
and locks access to the static list of SRTP sessions to prevent potential
race conditions.
BUG=4042
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/52609004
Cr-Commit-Position: refs/heads/master@{#9326}
ession/media/srtpfilter.cc
|
| 9e3cb336d4c62d4632552293eddd59612829ece1 |
29-May-2015 |
Alex Glaznev <glaznev@google.com> |
AppRTCDemo: check for necessary permissions before starting the call.
Also update PeerConnection.RTCConfiguration values.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56559004
Cr-Commit-Position: refs/heads/master@{#9325}
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 5ee9f679a550cc243ec94def5e4f321546bbf312 |
29-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove webrtcvideoengine.cc.
This file is no longer built here or in Chromium and can be removed.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54599004
Cr-Commit-Position: refs/heads/master@{#9322}
edia/webrtc/webrtcvideoengine.cc
|
| 7c4e7458b5ce99c13a75d5be7d718ef94e2f8f9f |
28-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Support multiple URLs in PeerConnectionInterface::IceServer
This adds support for multiple URLs in a IceServer configuration as
defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCIceServer.
BUG=2096
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57489004
Cr-Commit-Position: refs/heads/master@{#9320}
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface.h
|
| d4f769d8fc48769eff226392b1ae105161b3e7c4 |
28-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
Stop video candidates getting down to audio.
Second attempt at adding a check to make sure that the video
transportproxy doesn't send down candidates to the audio transport
channel when things are bundled.
BUG=4665
R=juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50059004
Cr-Commit-Position: refs/heads/master@{#9316}
pp/webrtc/webrtcsession_unittest.cc
|
| 259bd2034c3d3ee7f2dc4d481e9bf896a3a4d6ef |
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Report ssrc_groups in GetStats().
This was already available in the stats struct, just not filled in.
BUG=4720
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47329004
Cr-Commit-Position: refs/heads/master@{#9308}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 3b187b9c0c4f404c31523e74909ee3b75a83846e |
28-May-2015 |
Henrik Boström <hbos@webrtc.org> |
Removed unnecessary includes of webrtcvideocapturer.h
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57469004
Cr-Commit-Position: refs/heads/master@{#9305}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/videotrack.cc
ibjingle.gyp
edia/webrtc/webrtcvideoengine2.cc
|
| 23c2e5547904c633959b583ba6b50588f347581d |
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove remaining .mk files.
These files are not supported, kept up to date or likely to build
anymore.
BUG=
R=glaznev@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46489004
Cr-Commit-Position: refs/heads/master@{#9303}
pp/webrtc/androidtests/jni/Android.mk
xamples/android/jni/Android.mk
|
| fec2c6d7eb58574b32eaa26222d3fb903b738cfa |
27-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Prevent potential double-free if srtp_create fails.
If srtp_create fails while adding streams, it deallocates the session
but doesn't clear the passed pointer which then could lead to a
double-free in the SrtpSession dtor.
The CL also adds locking for libsrtp initialization / shutdown.
BUG=4042
R=jiayl@webrtc.org, juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47319004
Cr-Commit-Position: refs/heads/master@{#9300}
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
|
| cbe408aa118e46e1f1dd28d201378968f00b60ea |
27-May-2015 |
Henrik Boström <hbos@webrtc.org> |
WebRtcVideoCapturer: Getting rid of the |critical_section_stopping_| lock and all of its critical sections.
This avoids a deadlock in WebRtcVideoCapturer.
The deadlock could occur because OnIncomingFrame() has the |critical_section_stopping_| lock, which could block a Stop() on the |start_thread_|. When OnIncomingFrame() then tries to do synchronous invoke on |start_thread_| (before releasing said lock) we have a deadlock.
BUG=4670
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47259004
Cr-Commit-Position: refs/heads/master@{#9294}
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
|
| f09e09c7eef8722cc6902069a1ab7deb8948f98b |
26-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
VoE: Remove unused interfaces
BUG=4690
I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53519004
Cr-Commit-Position: refs/heads/master@{#9281}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 54be3e004981b1f3e0214d59f86bcfb3c3be9c33 |
25-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove some WebRtcVideoEngine2 unittest stubs.
Also contains some cleanup/typo fixes.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55449004
Cr-Commit-Position: refs/heads/master@{#9277}
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 0eefb4d5c3a11ec44360ca1fc144a5288fe4d6d0 |
23-May-2015 |
Tommi <tommi@chromium.org> |
Detach base/logging.* from base/stream.*.
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54529004
Cr-Commit-Position: refs/heads/master@{#9269}
pp/webrtc/java/jni/peerconnection_jni.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 469c2c04aae3e8446ba35f482adabd42800b41e1 |
23-May-2015 |
Andrew MacDonald <andrew@webrtc.org> |
Make Config::default_value leak instead of having an exit-time destructor.
I wanted to use Config::Get in Chromium code, but it triggered the following
warning:
../../third_party/webrtc/common.h:89:20: error: declaration requires an exit-time destructor [-Werror,-Wexit-time-destructors]
static const T def;
^
../../third_party/webrtc/common.h:110:10: note: in instantiation of function template specialization requested here
return default_value<T>();
^
I assume we don't hit this in webrtc because the warning is disabled.
This also switches to the RTC_ prefix from the deprecated LIBJINGLE_.
Needed due to this Chromium CL:
https://codereview.chromium.org/1148843004/
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53459004
Cr-Commit-Position: refs/heads/master@{#9268}
ession/media/srtpfilter.cc
|
| 4bf12eafba4e18504ac22080262dba61b4c8b9e7 |
23-May-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Fix sending wrong candidates down to transportchannel."
This reverts commit f65de8483e90d1d52d5d8f40f646e77bf45b10ea.
It was breaking the build bots: http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/3062
TBR=decurtis
BUG=
Review URL: https://webrtc-codereview.appspot.com/54539004
Cr-Commit-Position: refs/heads/master@{#9267}
pp/webrtc/webrtcsession_unittest.cc
|
| f65de8483e90d1d52d5d8f40f646e77bf45b10ea |
22-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
Fix sending wrong candidates down to transportchannel.
BUG=4665
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54489004
Cr-Commit-Position: refs/heads/master@{#9266}
pp/webrtc/webrtcsession_unittest.cc
|
| 3548dd21542c7b3f2c4680c6a6d86b0d719bd008 |
22-May-2015 |
Peter Boström <pbos@webrtc.org> |
Set local SSRCs on receivers added before senders.
Addresses bug where a receiver would report SSRC 1 even though the
endpoint has sending streams.
BUG=4678
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51099004
Cr-Commit-Position: refs/heads/master@{#9262}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 915df4fc308489bddda94cbfe38abba1d3fa4736 |
22-May-2015 |
Henrik Boström <hbos@webrtc.org> |
CaptureManager: Don't stop a capturer at UnregisterVideoCapturer if it did not start in the first place.
This fixes a bug where, if the VideoCapturer failed to start under certain circumstances, the capture manager would cause a callback saying that the capturer stopped even though it never started in the first place. A VERIFY check in VideoSource::SetState would then cause a crash since the state was set to kEnded when it was already in state kEnded (SetState only allows being called when the state changes).
I only noticed this bug while doing a mistake in a separate CL. Not sure how to reliably reproduce said bug on a working build, but I have previously had camera hardware issues where it couldn't start the camera which resulted in the SetState kEnded -> kEnded crash. Hopefully this will fix that.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51039004
Cr-Commit-Position: refs/heads/master@{#9259}
edia/base/capturemanager.cc
|
| 9a416bd14ee225d8f1a1ada627a1dd7daf275032 |
22-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51879004
Cr-Commit-Position: refs/heads/master@{#9258}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 98d8cf58ee8c7c7b0672ce7955313a31824d6f3a |
21-May-2015 |
jackychen <jackychen@webrtc.org> |
Hardware VP8 encoding: Use QP as metric for resize.
Add vp8 frame header parser to get QP from vp8 bitstream.
BUG= 4273
R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49259004
Cr-Commit-Position: refs/heads/master@{#9256}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| af55ccc054de9b91f6e5f5059937a91c0c91ff30 |
21-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add RtcpMuxPolicy support to PeerConnection.
BUG=4611
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/46169004
Cr-Commit-Position: refs/heads/master@{#9251}
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/rtcpmuxfilter.cc
ession/media/rtcpmuxfilter.h
ession/media/rtcpmuxfilter_unittest.cc
|
| 76b62ff1ad4819ab11133d30abafd705e78a387f |
20-May-2015 |
Tommi <tommi@webrtc.org> |
Clean up now-unused code that was used for libpeerconnection.[so|dll].
BUG=chromium:463660
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56409004
Cr-Commit-Position: refs/heads/master@{#9240}
uild/common.gypi
ibjingle.gyp
edia/webrtc/webrtcexport.h
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvoiceengine.h
|
| fce324272d0d9f6762b0b7d9f0c081b3692aa4ef |
20-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove linphonemediaengine.*
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54479004
Cr-Commit-Position: refs/heads/master@{#9239}
ibjingle.gyp
edia/other/linphonemediaengine.cc
edia/other/linphonemediaengine.h
|
| c3f4dbc40b9369a7f8eb9248adb8a018b9d8e439 |
20-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove rtp_rtcp/ dump functionality.
Removes RTP dumping from VideoEngine and VoiceEngine.
BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47179004
Cr-Commit-Position: refs/heads/master@{#9234}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 831c5585c7d2b4c4442e3c1255332f1c23b6a983 |
20-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Allow setting maximum protocol version for SSL stream adapters.
This CL adds an API to SSL stream adapters to set the maximum allowed
protocol version and with that implements support for DTLS 1.2.
With DTLS 1.2 the default cipher changes in the unittests as follows.
BoringSSL
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA -> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
NSS
TLS_RSA_WITH_AES_128_CBC_SHA -> TLS_RSA_WITH_AES_128_GCM_SHA256
BUG=chromium:428343
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/50989004
Cr-Commit-Position: refs/heads/master@{#9232}
pp/webrtc/peerconnection_unittest.cc
|
| 4d71edef45afa38b3f68b6af0519ac0f21d327df |
19-May-2015 |
Peter Boström <pbos@webrtc.org> |
Add HW fallback option to software encoding.
Permits falling back to software encoding for unsupported resolutions.
BUG=chromium:475116, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46279004
Cr-Commit-Position: refs/heads/master@{#9227}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 17b889b899c3eeaa96dc26a6e61cd2e6296b0fec |
19-May-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Issue 4366: Adapted frames have wrong width and height and are cropped.
When a frame being stretched, the original rotation information is lost. This is to ensure it's carried over.
Also removed StretchToBuffer function as it's not called and dangerous.
BUG=4366
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51869004
Cr-Commit-Position: refs/heads/master@{#9224}
ibjingle_tests.gyp
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframefactory_unittest.cc
|
| 2f5be9ad630dfe499a3e2fc64c6178143acddb84 |
19-May-2015 |
Alex Glaznev <glaznev@google.com> |
Improve Android camera error handling.
- Set Camera.ErrorCallback callback when opening camera to
receive camera server error notifications.
- Allow user to provide interface for handling camera errors
happening on camera thread.
- Run camera observer on camera thread and monitor camera fps
and amount of callback buffers, print statistics and report error
if camera stops generating frames.
- Query camera formats starting from front camera instead of back
camera to detect camera failures as fast as possible.
- Change all DCHECK to CHECK in androidvideocapturer.cc to detect
camera error on release builds.
- Plus adding some extra logging.
R=hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52519004
Cr-Commit-Position: refs/heads/master@{#9221}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| ccb49e79fd4c439a30b9a999eab4ef329ba8425c |
19-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51009004
Cr-Commit-Position: refs/heads/master@{#9216}
pp/webrtc/peerconnectionfactory.cc
ibjingle.gyp
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/other/linphonemediaengine.h
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/soundclip.cc
ession/media/soundclip.h
|
| b92be45c858b43d1d064f3c49796a440e9917874 |
19-May-2015 |
Weiyong Yao <braveyao@webrtc.org> |
Support 720P in portait as maximum on iOS.
BUG=4643
TEST=Manual Test and trybots
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53419004
Cr-Commit-Position: refs/heads/master@{#9214}
pp/webrtc/jsepsessiondescription.cc
|
| 2e7a09800595d4d82f67acfd7de04794642cef7d |
18-May-2015 |
Noah Richards <noahric@chromium.org> |
Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49989004
Cr-Commit-Position: refs/heads/master@{#9210}
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 7252a2ba8035c4128917a9558a3e34fc9dbe7c44 |
18-May-2015 |
Peter Boström <pbos@webrtc.org> |
Add HW fallback option to software decoding.
Permits falling back to software decoding for unsupported resolutions in
bitstreams.
BUG=4625, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46269004
Cr-Commit-Position: refs/heads/master@{#9209}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| b26198972c1fcb4aa7abaf3895b007e301e7d5dc |
18-May-2015 |
henrika <henrika@chromium.org> |
Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
ibjingle.gyp
|
| 144d01850bd3e07222d3f8696debec689dcdccf5 |
15-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
fix indent on tokenize_first function signatures
R=juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52499004
Cr-Commit-Position: refs/heads/master@{#9198}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 0e07f92043b333acfdaed8f22da5df903a70e0e9 |
15-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
Split fmtp on semicolons not spaces as per RFC6871
BUG=4617
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47169004
Cr-Commit-Position: refs/heads/master@{#9193}
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 4cd6940e49e1caed059e7cc7f43a1f9232725df4 |
15-May-2015 |
Henrik Kjellander <kjellander@chromium.org> |
Enable -Wformat-security warning and cleanup GYP.
Enable the -Wformat-security and -Wformat warnings for talk/.
Remove *.def and *.h.pump files from webrtc/base/base.gyp since they're not supported by some tools.
BUG=4242
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49969004
Cr-Commit-Position: refs/heads/master@{#9191}
uild/common.gypi
|
| 39f2b0c870c59dbc37d4f4d8cdb5fff7a7ae5b81 |
14-May-2015 |
Yuriy Shevchuk <youwrk@gmail.com> |
Implemented video device info for iOS
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42189004
Patch from Yuriy Shevchuk <youwrk@gmail.com>.
Cr-Commit-Position: refs/heads/master@{#9190}
edia/webrtc/webrtcvideocapturer.cc
|
| 2013aeced2b7821a407f302802c4a16fd02728b1 |
13-May-2015 |
Minyue <minyue@webrtc.org> |
Propagating RTT from send-only channel to receive-only channel.
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.
BUG=3978
TEST=chromium with hangout calls
R=henrika@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29989004
Cr-Commit-Position: refs/heads/master@{#9186}
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 300eeb68f55c5091c7045e377578586733cddf16 |
12-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoEngine interfaces.
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.
Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).
BUG=1695, 4491
R=mflodman@webrtc.org, solenberg@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55379004
Cr-Commit-Position: refs/heads/master@{#9179}
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvie.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 67c9df79828991c5aab96b9253ae4e7ba7ed508e |
11-May-2015 |
Peter Boström <pbos@webrtc.org> |
Base NACK on send codecs.
Addressing discrepancy where NACK used to be set from send codecs in
WebRtcVideoEngine(1), and before this change, from recv codecs in
WebRtcVideoEngine2. This should address that NACK might be sent even if
the remote side does not support it.
BUG=4626
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53409004
Cr-Commit-Position: refs/heads/master@{#9171}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 126c03ea02d8a99bfa3d1e6d6fe04516183d31af |
11-May-2015 |
Peter Boström <pbos@webrtc.org> |
Base decision to send REMB on send codecs.
Fixes bug where Chromium would send REMB even though the remote party
doesn't announce support for it (because it was based on local codec
settings instead of remote ones).
BUG=4626
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54389004
Cr-Commit-Position: refs/heads/master@{#9170}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 64dad838e61e92e4a72437b153c5eba7a200fb4a |
11-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49329004
Cr-Commit-Position: refs/heads/master@{#9169}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 1f629232d5f852452499104c28e7d61c7b0b8c77 |
10-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55369004
Cr-Commit-Position: refs/heads/master@{#9165}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| fd32f35aff8fc28ec084bddc274de284e0422a57 |
10-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.
Contains a tentative fix to the chrome build breakage caused by the
original change.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47139004
Cr-Commit-Position: refs/heads/master@{#9164}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 4c277bb938854a6a174e8bfece0bc9a7928da1ab |
08-May-2015 |
Lally Singh <lally@google.com> |
Add basic SCTP packet logging.
Attempts to get wireshark to decode the DTLS were problematic (wireshark only does it for certain versions of some DTLS implementations), so just do what firefox does and dump a txt2pcap-compatible log when requested.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49159004
Cr-Commit-Position: refs/heads/master@{#9162}
edia/sctp/sctpdataengine.cc
|
| cdb47a4533b7b1e29e803ed6591a68bb1a4f1692 |
08-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53399004
Cr-Commit-Position: refs/heads/master@{#9161}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 208a2294cde839025318f1b3d57559cb0611a4e7 |
08-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.
R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50869004
Cr-Commit-Position: refs/heads/master@{#9159}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| d3ddc1b69e9cdfd7c6d38ab02b8d8ab891d30fd1 |
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
BUG=
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49929004
Cr-Commit-Position: refs/heads/master@{#9156}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccommon.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| e444a3dcd317ff81b344a89625376e2afcffb1e2 |
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
WebRtcVoiceEngine: Get rid of unnecessary template base class.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46219004
Cr-Commit-Position: refs/heads/master@{#9155}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| aaf8ff2e45ece09028b8064eec6234260d9cc081 |
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
WebRtcVoiceEngine: virtual to override + git cl format.
BUG=
R=kwiberg@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54369004
Cr-Commit-Position: refs/heads/master@{#9154}
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 6179b89e53eda4db57baf2efb8d85779defb410c |
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove unused API on WebRtcVoiceEngine.
BUG=1695
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46209004
Cr-Commit-Position: refs/heads/master@{#9153}
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 4b60c73e74d62beff484b7f54d8f3267cb66274f |
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49269004
Cr-Commit-Position: refs/heads/master@{#9150}
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediachannel.h
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 81ea54eaac82b36b7208a02fd37a469d7d0bd9d0 |
07-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove WebRtcVideoEngine.
Leaves a stub file for talk/media/webrtc/webrtcvideoengine.cc until
build files in Chromium have been modified.
BUG=1695,4566
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48339004
Cr-Commit-Position: refs/heads/master@{#9148}
ibjingle.gyp
ibjingle_tests.gyp
edia/base/mediaengine.cc
edia/webrtc/constants.h
edia/webrtc/dummyinstantiation.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| ccfc93913ce015309429ea07ddf24808f111efb9 |
07-May-2015 |
Bjorn Volcker <bjornv@chromium.org> |
Reinterpret AudioOption delay_agnostic_aec to override HW-AEC
This CL will change the behavior when enabling Delay Agnostic AEC through the media constraint (and AudioOption delay_agnostic_aec)
FROM
Use DA-AEC instead of AECM if there is no HW-AEC
TO
Use DA-AEC even if there is a HW-AEC
Before this change the user will not really know if the Delay Agnostic AEC is running or not, so it is more intuitive if the option overrides the built-in one if the user has asked for it.
BUG=4472
TESTED=locally with a modified AppRTCDemo app
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49859004
Cr-Commit-Position: refs/heads/master@{#9147}
edia/webrtc/webrtcvoiceengine.cc
|
| 57cc74e32cb85dc33bd93c1c4a3609a4a090b244 |
05-May-2015 |
Zeke Chin <tkchin@webrtc.org> |
iOS camera switching video capturer.
Introduces a new capture class derived from cricket::VideoCapturer that
provides the ability to switch cameras and updates AppRTCDemo to use it.
Some future work pending to clean up AppRTCDemo UI.
BUG=4070
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48279005
Cr-Commit-Position: refs/heads/master@{#9137}
pp/webrtc/objc/RTCAVFoundationVideoSource+Internal.h
pp/webrtc/objc/RTCAVFoundationVideoSource.mm
pp/webrtc/objc/RTCPeerConnectionFactory+Internal.h
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCVideoTrack.mm
pp/webrtc/objc/avfoundationvideocapturer.h
pp/webrtc/objc/avfoundationvideocapturer.mm
pp/webrtc/objc/public/RTCAVFoundationVideoSource.h
pp/webrtc/objc/public/RTCVideoTrack.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
xamples/objc/AppRTCDemo/ios/resources/ic_switch_video_black_24dp.png
xamples/objc/AppRTCDemo/ios/resources/ic_switch_video_black_24dp@2x.png
ibjingle.gyp
ibjingle_examples.gyp
|
| c56ac1ec298630ba95e44a9da9efeb9d1a6d43d4 |
04-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer: Remove backwards compatibility band-aids
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
(This was previously committed (9e1a6d7c) but had to be reverted
(cbf09274) because Chromium on Android wasn't quite ready for it).
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47109004
Cr-Commit-Position: refs/heads/master@{#9132}
ession/media/channel.cc
|
| e433c0ef31297d78336d99cc18cf063b1a486cf2 |
01-May-2015 |
Alex Glaznev <glaznev@google.com> |
Restore back verbosity logging for camera captured frame.
Helps to debug camera freezes.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46179004
Cr-Commit-Position: refs/heads/master@{#9127}
edia/webrtc/webrtcvideoengine2.cc
|
| cac1b381359652fe623a1bdd320f0877b7b7a300 |
30-Apr-2015 |
Jiayang Liu <jiayl@chromium.org> |
Expose RTCConfiguration to java JNI and add an option to disable TCP
BUG=4585, 4589
R=glaznev@webrtc.org, juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49809004
Cr-Commit-Position: refs/heads/master@{#9125}
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 4eddf18b1c4a93bc9c736783094cc4204c2d955e |
30-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle.
BUG=
R=decurtis@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/46149004
Cr-Commit-Position: refs/heads/master@{#9124}
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| cbf0927473c10a0a25bbf55707f1ca2b2fd57708 |
30-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Revert "rtc::Buffer: Remove backwards compatibility band-aids"
This reverts commit 9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07, because
Chromium for Android still isn't happy with it.
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49869004
Cr-Commit-Position: refs/heads/master@{#9122}
ession/media/channel.cc
|
| 9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07 |
30-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer: Remove backwards compatibility band-aids
This CL makes two changes to rtc::Buffer that have had to wait for
Chromium's use of it to be modernized:
1. Change default return type of rtc::Buffer::data() from char* to
uint8_t*. uint8_t is a more natural type for bytes, and won't
accidentally convert to a string. (Chromium previously expected
the default return type to be char, which is why
rtc::Buffer::data() initially got char as default return type in
9478437f, but that's been fixed now.)
2. Stop accepting void* inputs in constructors and methods. While
this is convenient, it's also dangerous since any pointer type
will implicitly convert to void*.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44269004
Cr-Commit-Position: refs/heads/master@{#9121}
ession/media/channel.cc
|
| f16fcbec734e1e3303828525c9fd7e13e0803aab |
30-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViECapture usage in VideoSendStream.
Instead a ViECapturer object is allocated and directly operated on. This
additionally exposes ViESharedData to Call to access the module
ProcessThread, moving towards Call ownership of shared resources.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45339004
Cr-Commit-Position: refs/heads/master@{#9119}
edia/webrtc/fakewebrtcvideoengine.h
|
| efbde3775b5eed8015d7e2e86ddcea3e6033d321 |
29-Apr-2015 |
Erik Språng <sprang@webrtc.org> |
Don't use CPU adaptation for screen content in the new API.
BUG=4605
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48309004
Cr-Commit-Position: refs/heads/master@{#9116}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| adf89b7e33cc54dab9365dddead687a46a074cf0 |
29-Apr-2015 |
Ivo Creusen <ivoc@webrtc.org> |
Added SetBitRate function to VoE API to allow changing the audio bitrate.
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.
BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50789004
Cr-Commit-Position: refs/heads/master@{#9115}
edia/webrtc/fakewebrtcvoiceengine.h
|
| 23fba1ffa0079f70744a83bcf4e85501dc226013 |
29-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51749004
Cr-Commit-Position: refs/heads/master@{#9114}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2.cc
|
| 10ba3eec5a268d359260e331fcc2608f818d5236 |
29-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Roll chromium_revision a12e1e1..0cb2549 (326495:327252)
https://codereview.chromium.org/1051343002 adds a dependency
on Chromium's third_party/junit into base/ which affects our
Android tests that uses that code.
The precompiled JUnit 4.11 JAR file that is only by the
libjingle_peerconnection_java_unittest target on Linux has been
moved to third_party/junit-jar, since it collided with the expected
path for the JUnit dependency mentioned above.
It had to be kept since the Chromium JUnit is only possible to build
when OS==android.
This CL also brings in Mockito and Robolectric, which should be
useful for our Android tests.
Other relevant changes:
* src/buildtools: 3b302fe..15308f4
* src/third_party/libjpeg_turbo: 034e9a9..9e9058b
* src/third_party/libyuv: 32ad6e0..01db3d1
Details: https://chromium.googlesource.com/chromium/src/+/a12e1e1..0cb2549/DEPS
Clang version was not updated in this roll.
BUG=4499
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48239004
Cr-Commit-Position: refs/heads/master@{#9113}
ibjingle_tests.gyp
|
| 94cc1fe4af57a01a99a1f76f0ad3d48edf981321 |
29-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViEImageProcess usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also removing some methods from ViEImageProcess
that were only added for Video{Send,Receive}Stream usage.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45319004
Cr-Commit-Position: refs/heads/master@{#9111}
edia/webrtc/fakewebrtcvideoengine.h
|
| 1ba344a07060bf57db649299835ec4f093d58d40 |
29-Apr-2015 |
Bjorn Volcker <bjornv@chromium.org> |
Adds a MediaConstraint for the AudioOption aec_dump
Alson includes
- a test verifying that the option is set
- changed the test verifying delay_agnostic_aec option is set to use non-default value
BUG=4555
TESTED=locally through AppRTCDemo on N7 and Android One
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46059004
Cr-Commit-Position: refs/heads/master@{#9109}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
|
| faa6d076b7d021467cc05e7c293940773524cad8 |
28-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Remove a few verbose log messages from webrtcvideoengine2.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49189004
Cr-Commit-Position: refs/heads/master@{#9105}
edia/webrtc/webrtcvideoengine2.cc
|
| 019087f5bb294d9590b4ed68347ad79d9335ad74 |
28-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add safeguards against signalling peer-reflexive candidates.
BUG=4208
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/50799004
Cr-Commit-Position: refs/heads/master@{#9104}
pp/webrtc/webrtcsdp.cc
|
| 143cec1cc68b9ba44f3ef4467f1422704f2395f0 |
28-Apr-2015 |
Erik Språng <sprang@google.com> |
Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.
BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46069004
Cr-Commit-Position: refs/heads/master@{#9093}
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/simulcast.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| e6cefb60f84b96ab6a06573cec68a836ab0eec81 |
27-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
GYP variables for building expat, icu, libsrtp, usrsctp
This makes the build more flexible when linking against
prebuilt external libraries.
Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).
Also make it possible to avoid building the GTK parts of the Linux build.
BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44179005
Cr-Commit-Position: refs/heads/master@{#9087}
pp/webrtc/java/jni/jni_helpers.cc
uild/common.gypi
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| 77d444a433d616adf3c83fa1b37098856b0c8e52 |
24-Apr-2015 |
Tommi <tommi@webrtc.org> |
Handle the case when hoststring is empty.
BUG=chromium:480536
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46109004
Cr-Commit-Position: refs/heads/master@{#9081}
pp/webrtc/peerconnection.cc
|
| c4188fd3c74688264621393fc622cb81c042c1ac |
24-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Use IncomingVideoStream in VideoReceiveStream.
Decouples VideoReceiveStream further from webrtc/video_engine/ as well
as most of webrtc/modules/video_render/ resulting in a simpler setup.
BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50749004
Cr-Commit-Position: refs/heads/master@{#9080}
edia/webrtc/fakewebrtcvideoengine.h
|
| 24d448561403358d32958628a5ef54b1cd416857 |
23-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Enable -Wunused-private-field warning for talk/
BUG=4242
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49139004
Cr-Commit-Position: refs/heads/master@{#9069}
uild/common.gypi
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcvideoengine.cc
|
| 352595459dbd8d0ea63ab6241204340d917c9739 |
23-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Use short include paths for icu headers.
This makes it possible to build with icu located
in another absolute path.
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46079004
Cr-Commit-Position: refs/heads/master@{#9063}
pp/webrtc/java/jni/jni_helpers.cc
pp/webrtc/java/jni/jni_helpers.h
pp/webrtc/java/jni/peerconnection_jni.cc
|
| ee0b00e8a9cc2d8f4578912a389dee92ac020ee9 |
22-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Prevent recv-stream reconfig on identical codecs.
Receive streams seem to be reconfigured with identical codecs when
another stream is removed. Preventing this reconfiguration makes sure
that existing streams don't report stats during teardown when the stream
is still supposed to be running.
BUG=1788
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44249004
Cr-Commit-Position: refs/heads/master@{#9059}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 908e77bd00593a8bc5480fb1a07368b7c27778f3 |
22-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Allow Java code to detect if VP8 and H.264 HW decoding is supported.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43269004
Cr-Commit-Position: refs/heads/master@{#9058}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| b67288283a7e727200efca2b578e446a1a1c4225 |
22-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Move cricket::FakeCall and associates to a separate file.
BUG=4574
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49129004
Cr-Commit-Position: refs/heads/master@{#9057}
ibjingle_tests.gyp
edia/webrtc/fakewebrtccall.cc
edia/webrtc/fakewebrtccall.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 7fb711f68312f61f392b3f33b950e97cb07da71f |
22-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class.
BUG=4574
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50769004
Cr-Commit-Position: refs/heads/master@{#9056}
pp/webrtc/statscollector_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/mediarecorder_unittest.cc
|
| 393347ff988708df5037ddcd181fe204bd1ab37e |
22-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Report receive-side packet loss.
BUG=4558
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48219004
Cr-Commit-Position: refs/heads/master@{#9054}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 7c027b64ae53a29bc528b4241cc540694c239304 |
22-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Enable more Clang warnings for talk/
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46999004
Cr-Commit-Position: refs/heads/master@{#9053}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/statscollector_unittest.cc
uild/common.gypi
edia/sctp/sctpdataengine.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvoiceengine.cc
ession/media/channel.cc
|
| 61b4d518af7c2e4156056931d3512a49032b827d |
22-Apr-2015 |
jackychen <jackychen@webrtc.org> |
Dynamic resolution change for VP8 HW encode.
Off by default for now.
BUG=
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45849004
Cr-Commit-Position: refs/heads/master@{#9045}
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| e62202fedf57b74cc263246c0586ee353978caf8 |
21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.
Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.
BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36889004
Cr-Commit-Position: refs/heads/master@{#9040}
edia/base/codec.cc
edia/base/codec.h
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| c4905fb72a01d5fe5788cc33d847c31b039468e3 |
21-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Fix race condition in Android camera JNI code.
AndroidVideoCapturerJni dtor is called on signaling thread
and may destroy JNI global refs while processing late camera
frame arrival in ReturnBuffer_w() in worker thread.
Fix this by waiting for all function invoked on worker thread
to complete in camera JNI dtor.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49099004
Cr-Commit-Position: refs/heads/master@{#9037}
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
|
| ac7d97fea6a6781ad3a77c8653e62ea9ec1f188c |
20-Apr-2015 |
Zeke Chin <tkchin@webrtc.org> |
Remove frame copy in RTCOpenGLVideoRenderer.
BUG=1128
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44039004
Cr-Commit-Position: refs/heads/master@{#9036}
pp/webrtc/objc/RTCEAGLVideoView.m
pp/webrtc/objc/RTCOpenGLVideoRenderer.mm
|
| 8c054154daa17676ad32ca6554025b7a09670410 |
20-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Add extra logging for Android camera JNI layer.
Plus enabled checks for release version.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46939004
Cr-Commit-Position: refs/heads/master@{#9034}
pp/webrtc/java/jni/androidvideocapturer_jni.cc
|
| 9478437fdea4eb31b92ffe0c10368fe5bc9b9e16 |
20-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
int8_t*, and char*. Previously, they accepted void*, meaning that
any kind of pointer was accepted. I think requiring an explicit
cast in cases where the input array isn't already of a byte-sized
type is a better compromise between convenience and safety.
2. data() can now return a uint8_t* instead of a char*, which seems
more appropriate for a byte array, and is harder to mix up with
zero-terminated C strings. data<int8_t>() is also available so
that callers that want that type instead won't have to cast, as
is data<char>() (which remains the default until all existing
callers have been fixed).
3. Constructors, SetData(), and AppendData() now accept arrays
natively, not just decayed to pointers. The advantage of this is
that callers don't have to pass the size separately.
4. There are new constructors that allow setting size and capacity
without initializing the array. Previously, this had to be done
separately after construction.
5. Instead of TransferTo(), Buffer now supports swap(), and move
construction and assignment, and has a Pass() method that works
just like std::move(). (The Pass method is modeled after
scoped_ptr::Pass().)
R=jmarusic@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42989004
Cr-Commit-Position: refs/heads/master@{#9033}
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/sctputils.cc
pp/webrtc/test/mockpeerconnectionobservers.h
edia/base/fakemediaengine.h
edia/base/filemediaengine.cc
edia/base/rtpdataengine.cc
edia/base/rtpdataengine_unittest.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/srtpfilter.cc
|
| 91543731c3a850dcc52ae63be8cc257e507fb72d |
20-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Do not define POSIX.
It breaks integration with upstream re2 library on Chromium.
Without patching re2 library, with this define, it produces the
following error:
../../third_party/re2/re2/re2.h:254:5: error: expected identifier
POSIX, // POSIX syntax, leftmost-longest match
As we define POSIX on the command line, the C preprocessor changes
RE2::POSIX to nothing and thus break the compilation. :(
See chromium-dev mailing list for this discussion in
https://groups.google.com/a/chromium.org/d/topic/chromium-dev/UXCHnX7pV44/discussion
BUG=None
TEST=ninja -C out/Debug, everything compiles as before
R=sergeyu@chromium.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46049004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#9032}
uild/common.gypi
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/server/data_socket.cc
xamples/stunserver/stunserver_main.cc
|
| 09a9ea888620a683c891ce3e67bfaa40cc8dc6c2 |
17-Apr-2015 |
Henrik Boström <hbos@webrtc.org> |
Supporting formats of non-multiple of 16 widths on Android.
This is an updated version of perkj's issue (https://webrtc-codereview.appspot.com/44129004/) which was reverted due to libjingle_peerconnection_android_unittest crashing on Nexus 9. It crashed because there was old test code still assuming the width was multiple of 16 (which was only a problem on devices with non-16 widths).
BUG=4522
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45109004
Cr-Commit-Position: refs/heads/master@{#9029}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| f49dbfa5c3fab0b26dc4d3a139993a2fe7a1c735 |
16-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Close all camera resources when camera error happens.
Also add more logs to better track still observed camera
open/close failures.
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48109004
Cr-Commit-Position: refs/heads/master@{#9020}
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 9829af4bfc8f86f5fd9626b83bc3a22a05efb986 |
15-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Disable VP8 encoder HW acceleration on NVidia devices.
NVidia HW encoder bitrate control is allowing too much
bitrate fluctuation. Plus average encoding time is not enough
for 720p 30 fps support.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48099004
Cr-Commit-Position: refs/heads/master@{#9014}
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 352b2d7a19d6313273608c26edf8900e592a3831 |
15-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream).
Add separate functions for returning stats from send/receive stream and updated how functions are used.
Add test implementation for histogram methods in system_wrappers/interface/metrics.h.
BUG=4519
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49639004
Cr-Commit-Position: refs/heads/master@{#9009}
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| 4b76c023625f717c0bffe9ab0d584f2795317f1d |
15-Apr-2015 |
Magnus Jedvert <magjed@google.com> |
Roll chromium_revision 8af41b3..dcb0929 (324854:325030)
This is a major libyuv update (almost 200 revisions):
https://chromium.googlesource.com/external/libyuv/+log/d204db6..32ad6e0
Relevant changes:
* src/third_party/libyuv: d204db6..32ad6e0
* src/third_party/nss: d1edb68..9506806
Details: https://chromium.googlesource.com/chromium/src/+/8af41b3..dcb0929/DEPS
Since bayer and Q420 format support have been removed from libyuv, all tests related to those format are removed.
Clang version was not updated in this roll.
R=kjellander@webrtc.org
TBR=tommi
Review URL: https://webrtc-codereview.appspot.com/48989004
Cr-Commit-Position: refs/heads/master@{#9008}
edia/base/videocommon.h
edia/base/videoframe.cc
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe_unittest.cc
ession/media/planarfunctions_unittest.cc
|
| 3c3f6460646183914629e5dab8ae5fcede4f0e80 |
15-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Prevent null-stream reconfigs on RTP extensions.
If a codec fails to set (e.g. there's no codec configured), this
prevents a stream reconfigure with an invalid config. Reconfiguring a
stream without correct codec settings causes a CHECK failure.
BUG=chromium:475116
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44219004
Cr-Commit-Position: refs/heads/master@{#9007}
edia/webrtc/webrtcvideoengine2.cc
|
| e432800aeb6b695bda14acf2d60c0200803b5218 |
14-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Enable CPU adaptation by default.
WebRtcVideoEngine2 doesn't support CPU-monitor-based adaptation and as
such requires encoder-time-based CPU adaptation to perform any
adaptation at all.
BUG=4536
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49679004
Cr-Commit-Position: refs/heads/master@{#9001}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 56d50288e0f5df75cddc3798b8e01cdb75f25c92 |
14-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove SignalCaptureStateChange from MediaEngine.
It's no longer used by anything.
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/48069004
Cr-Commit-Position: refs/heads/master@{#8994}
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.h
ession/media/channelmanager.cc
|
| 575a8024bc3a2591c2090c3a424aae10db306577 |
14-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Add an option to update mirror flag in Android video renderer.
Plus fixing incorrect mirror matrix for 90 and
270 degree rotations.
BUG=4398
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50689004
Cr-Commit-Position: refs/heads/master@{#8993}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/src/org/appspot/apprtc/CallActivity.java
|
| 1b67795dc255490c70078dcc424ca2a037742287 |
13-Apr-2015 |
Zeke Chin <tkchin@webrtc.org> |
Add i386 to ios fat library build script and use boringssl.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48839005
Cr-Commit-Position: refs/heads/master@{#8992}
uild/build_ios_libs.sh
|
| 77f0e3f7b6a0664661dc295eb235c543b8091554 |
13-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GetStartCaptureFormat and some related code.
It is no longer used by anything.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48039004
Cr-Commit-Position: refs/heads/master@{#8990}
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| e7b221f4760af10e29cb4c501e758cc3518f628b |
13-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove deadlock in WebRtcVideoEngine2.
Acquiring stream_lock_ in WebRtcVideoChannel2 in a callback from Call
forms a lock-order inversion between process-thread locks and libjingle
locks, manifesting as CPU adaptation requests blocking on stream
creation that is blocked on the CPU adaptation request finishing.
R=asapersson@webrtc.org, mflodman@webrtc.org
BUG=4535,chromium:475065
Review URL: https://webrtc-codereview.appspot.com/50679004
Cr-Commit-Position: refs/heads/master@{#8985}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| eba964f472ab28f29363c253e7ccb872d7995961 |
11-Apr-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
Revert "Support none multiple of 16 pixels width on android."
Buildbot Android Tests (L Nexus9)(dbg) consistently fails on Instrumentation test libjingle_peerconnection_android_unittest (VideoCapturerAndroidTest) after this CL was landed.
This reverts commit f4acf46c863f2d516b09b00b39608de7e506ac65.
BUG=
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45079004
Cr-Commit-Position: refs/heads/master@{#8981}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 99c2fe5d2b3f0dcc9d8c723b99ddd73648e73b82 |
10-Apr-2015 |
Noah Richards <noahric@chromium.org> |
Fix NullVideoEngine's CreateChannel implementation.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44149004
Cr-Commit-Position: refs/heads/master@{#8980}
edia/base/mediaengine.h
|
| e4ae8d855892f16baac54ac6d269da1ccef89eaf |
10-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Changes in VideoCapturerAndroid.
- Do not handle more than one camera switch request at a time
to avoid blocking camera thread with multiple switch requests.
- Add a callback to notify when camera switch has been done.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46859004
Cr-Commit-Position: refs/heads/master@{#8978}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| f4acf46c863f2d516b09b00b39608de7e506ac65 |
10-Apr-2015 |
Per <perkj@chromium.org> |
Support none multiple of 16 pixels width on android.
BUG=4522
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44129004
Cr-Commit-Position: refs/heads/master@{#8977}
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| a125d7d7adc5fe6147b07be06f2380a772b30bdd |
10-Apr-2015 |
henrika <henrika@chromium.org> |
Changes default audio mode in AppRTCDemo to MODE_RINGTONE.
Also prevents that we try to restore audio mode when it has not been changed.
TBR=glaznev
BUG=NONE
TEST=AppRTCDemo and verify that volume control switches from "Ringtone to Phone" mode when call starts and switches back to Ringtone mode when call ends.
Review URL: https://webrtc-codereview.appspot.com/46879004
Cr-Commit-Position: refs/heads/master@{#8975}
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
|
| 9bfe3daf7349b62647997ced9389baa8ab043afe |
10-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Remove i420_video_frame.h header.
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.
This should fix pbos' TODO in i420_video_frame.h.
Tested on Linux with the following command lines:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug
BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46819004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8973}
edia/webrtc/webrtcvideoengine2.h
|
| f6c003eda5beb9105415ddd824df8ce3e87eeeb3 |
10-Apr-2015 |
Magnus Jedvert <magjed@webrtc.org> |
cricket::VideoFrameFactory: Handle if created frame is null
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46869004
Cr-Commit-Position: refs/heads/master@{#8972}
edia/base/videoframefactory.cc
|
| 0184057d540b20009aabf4a0a70386ac4b426c47 |
10-Apr-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoAdapterTest: Replace FileVideoCapturer with FakeVideoCapturer
The unittests are currently flaky due to the use of FileVideoCapturer.
BUG=4317
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49649004
Cr-Commit-Position: refs/heads/master@{#8969}
edia/base/videoadapter_unittest.cc
|
| 76c53d36bc455fe89ca1f860d5171633198fe907 |
09-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViE interface usage from VideoReceiveStream.
References channels and underlying objects directly instead of using
interfaces referenced with channel id. Channel creation is still done as
before for now.
BUG=1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46849004
Cr-Commit-Position: refs/heads/master@{#8958}
edia/webrtc/fakewebrtcvideoengine.h
|
| 15cf019a0071c7b67af047ae826d6bb5fb8ca89e |
09-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Add field-trial flag to disable WebRtcVideoEngine2.
BUG=chromium:475164
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45059004
Cr-Commit-Position: refs/heads/master@{#8957}
edia/webrtc/webrtcmediaengine.cc
|
| 9b3f56ea055934a5d5416db0386c857494410acc |
09-Apr-2015 |
Per <perkj@chromium.org> |
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
This reverts commit e41d774c4d0a60066866fc2d0ae48dd0e839ff23.
Original code review: https://webrtc-codereview.appspot.com/43999004/
Reason for reland: There was nothing wrong with this cl as is, but it breaks chrome compatibility. We will now reland this and fix Chrome during roll.
Patset 1: Original cl.
Patchset 2: Removed more code that is no longer needed.
R=magjed@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org
BUG=1128
Review URL: https://webrtc-codereview.appspot.com/45049004
Cr-Commit-Position: refs/heads/master@{#8956}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| d61ebda94168e9057cfb3e27778401eafe52f163 |
08-Apr-2015 |
Jiayang Liu <jiayl@chromium.org> |
Fix the sigslot type of DtlsIdentityStore::WorkerTask.
BUG=4516
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49619004
Cr-Commit-Position: refs/heads/master@{#8954}
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
|
| ad1f9b61a3107ca27ee023990dc576abc38f05ac |
08-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove warning on input frames before config.
Removes log spam for AppRTC when only one client is connected.
BUG=4512
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48019005
Cr-Commit-Position: refs/heads/master@{#8947}
edia/webrtc/webrtcvideoengine2.cc
|
| 9e420afefcb107ac07ffbad283765f71bceb8ea5 |
07-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Fix potential race conditions in Android video renderer.
- Check texture properties update flag using the same lock under which
the flag value is set.
- Adjust texture properties inside frame queue lock.
- Plus adding extra logging to track video renderer properties updates.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45929004
Cr-Commit-Position: refs/heads/master@{#8941}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| e41d774c4d0a60066866fc2d0ae48dd0e839ff23 |
07-Apr-2015 |
Per <perkj@chromium.org> |
Revert "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."
This reverts commit 75db8612588b4fabdf1b05f4ab145f7737093b45.
Revert "Fix build breakage in WrappedI420Buffer::native_handle()"
This reverts commit 3211934ebf7cac3e6df2cb4aacb6e47cc1cffe2b.
Reason for revert: Breaks chrome build and tests on clank, See https://codereview.chromium.org/1067803002/
BUG=1128
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43079004
Cr-Commit-Position: refs/heads/master@{#8940}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 1d83f1e89f3e54b38d49ff877c763d0ac52fdb8b |
07-Apr-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
talk/media/webrtc/webrtcvoiceengine: Delay Agnostic AEC should not override HW-AEC
In https://webrtc-codereview.appspot.com/48699004/ I made the audio option delay_agnostic_aec override HW-AEC if such exists. That is not an expected behavior and is fixed in this CL.
In addition we now check if EnableBuiltInAEC() was successful before disabling the SW-AEC. This revealed a bug in that return value, also fixed here.
BUG=4472
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47969004
Cr-Commit-Position: refs/heads/master@{#8936}
edia/webrtc/webrtcvoiceengine.cc
|
| 49a862ec4cba4060f9897ceb24b215d316bc0544 |
07-Apr-2015 |
Per <perkj@chromium.org> |
Return pending buffers to Java VideoCapturerAndroid if camera is stopping
BUG=4510
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45009005
Cr-Commit-Position: refs/heads/master@{#8935}
pp/webrtc/java/jni/androidvideocapturer_jni.cc
|
| 75db8612588b4fabdf1b05f4ab145f7737093b45 |
07-Apr-2015 |
Per <perkj@chromium.org> |
Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.
BUG=1128
R=magjed@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43999004
Cr-Commit-Position: refs/heads/master@{#8932}
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| e095148869f1811471ad2ee4ceadba2e9f268fae |
06-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Port some fixes in AppRTCDemo.
- Make PeerConnectionClient a singleton.
- Fix crash in CpuMonitor.
- Remove reading constraints from room response.
- Catch and report camera errors.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43059004
Cr-Commit-Position: refs/heads/master@{#8930}
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/CpuMonitor.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| ef88309a6e2b3193cf1658bf245de295900ba4fe |
06-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Forward declare AudioFrame type in voiceprocess.h
No need to include this header since the API is just taking a pointer to
it.
BUG=1092
TEST=./webrtc/build/gyp_webrtc && ninja -C out/Debug
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44059004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8928}
edia/base/voiceprocessor.h
edia/webrtc/webrtcvoiceengine.cc
|
| 3354419a2d53c3bfee8077cfbb0d86022b03e94a |
02-Apr-2015 |
Per <perkj@chromium.org> |
Zero copy AndroidVideeCapturer.
This cl uses the YV12 buffers from Java without a copy if no rotation is needed. Buffers are returned to the camera when the encoder and renderers no longer needs them.
This add a new frame type WrappedI420Buffer based in that allows for wrapping existing memory buffers and getting a notification when it is no longer used.
AndroidVideoCapturer::FrameFactory::CreateAliasedFrame wraps frame received from Java. For each wrapped frame a new reference to AndroidVideoCapturerDelegate is held to ensure that the delegate can not be destroyed until all frames have been returned.
Some overlap exist in webrtcvideoframe.cc and webrtcvideengine.cc with https://webrtc-codereview.appspot.com/47399004/ that is expected to be landed before this cl.
BUG=1128
R=glaznev@webrtc.org, magjed@webrtc.org
TBR=mflodman@webrtc.org // For changes in webrtc/common_video/video_frame_buffer
Review URL: https://webrtc-codereview.appspot.com/49459004
Cr-Commit-Position: refs/heads/master@{#8923}
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 037bad7497ec5a07e7d3bc0f14c40927cef33826 |
02-Apr-2015 |
Henrik Boström <hbos@webrtc.org> |
~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug.
BUG=chromium:320200
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49579004
Cr-Commit-Position: refs/heads/master@{#8922}
edia/base/capturemanager.cc
|
| cb76b895728c888c60cf474070fff6aee4f8731c |
02-Apr-2015 |
Thiago Farina <tfarina@chromium.org> |
Cleanup: Move json.h into rtc namespace.
This should fix the TODO in that header.
BUG=None
TEST=ninja -C out/Debug still compiles everything.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47919004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8921}
xamples/peerconnection/client/conductor.cc
|
| 64c1e8cda5cb4db85c5c296bf2f6a8181af7de9d |
02-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8917}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
edia/base/videocapturer.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 722ef1fb59878dbf6d05b3492bd41d9a66d68620 |
01-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Remove henrike@ from OWNERS
Since he has left the team.
R=henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48789004
Cr-Commit-Position: refs/heads/master@{#8913}
WNERS
|
| 31331cfd2d3d17958942b67190c8b943c05b084f |
01-Apr-2015 |
Minyue <minyue@webrtc.org> |
Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad16de57053bb6aa8a916079e0534bdae.
Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.
Review URL: https://webrtc-codereview.appspot.com/47909004
Cr-Commit-Position: refs/heads/master@{#8911}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
edia/base/videocapturer.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 1b1c15cad16de57053bb6aa8a916079e0534bdae |
01-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8905}
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
edia/base/videocapturer.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 4b3c0d6f34c379c2a06f654b62403876b20180c6 |
01-Apr-2015 |
Jiayang Liu <jiayl@chromium.org> |
Use WebRTC API to convert byteorder in srtpfilter.
This CL uses WebRTC API to convert 64bit from big-endian to host-endian,
so the internal "be64_to_cpu" of libsrtp is not used. The code path of
"be64_to_cpu" in newer versions of libsrtp depends on compile-time
defines that are not available in WebRTC.
BUG=https://code.google.com/p/chromium/issues/detail?id=328475
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46749004
Cr-Commit-Position: refs/heads/master@{#8904}
ession/media/srtpfilter.cc
ession/media/srtpfilter_unittest.cc
|
| 4825356620887946f289cb16f2095878be04a125 |
31-Mar-2015 |
Zeke Chin <tkchin@webrtc.org> |
RTCDataChannel: Unregister data channel observer on dealloc.
BUG=4490
R=haysc@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45889004
Cr-Commit-Position: refs/heads/master@{#8903}
pp/webrtc/objc/RTCDataChannel.mm
|
| 379069f676c13443398a72f7da8d118c973a0809 |
31-Mar-2015 |
Magnus Jedvert <magjed@webrtc.org> |
VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const.
RenderFrame should not modify the I420VideoFrame (and we don't).
This CL changes the declaration of RenderFrame from:
int32_t RenderFrame(const uint32_t streamId, I420VideoFrame& videoFrame)
to:
int32_t RenderFrame(const uint32_t streamId, const I420VideoFrame& videoFrame)
BUG=1128
R=mflodman@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46689005
Cr-Commit-Position: refs/heads/master@{#8902}
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcpassthroughrender_unittest.cc
|
| 23914fe756903353eae13fffc868d2c84f51f06f |
31-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject RTP one-byte extension ID 0.
Only accept local identifiers in the range 1-14 inclusive.
BUG=1788, chromium:471328
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50549004
Cr-Commit-Position: refs/heads/master@{#8900}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 0194d32873a5984d6aee1924404293dd169de903 |
30-Mar-2015 |
Alex Glaznev <glaznev@google.com> |
Add WebRtcAudioManager to peerconnection_jar library
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42969004
Cr-Commit-Position: refs/heads/master@{#8896}
ibjingle.gyp
|
| 1ecfd55044ca85f0564126382e33871260f574be |
30-Mar-2015 |
Magnus Jedvert <magjed@webrtc.org> |
videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)'
This CL reverts some parts of "Delete VideoAdapter::AdaptFrame" https://webrtc-codereview.appspot.com/44769004/.
Reason for revert: Should not touch HAVE_WEBRTC_VIDEO since libjingle_media_unittests does not compile without anyway.
BUG=4317
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48699005
Cr-Commit-Position: refs/heads/master@{#8888}
edia/base/videoadapter_unittest.cc
|
| dfd53fe26b013d0948024a38eec6fbc31c29a094 |
27-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Raise streams for SetMaxSendBitrates above 2000k.
Fixes b=AS effectively not setting bitrates above 2000k.
BUG=1788,4469
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47839004
Cr-Commit-Position: refs/heads/master@{#8882}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 53eda3dbd02a428178e7f9f40d2a4375c779cca8 |
27-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Add tests for r8811.
All these tests crashed before r8811. These tests should've been with
that change but r8811 was pushed in before to make bots green.
BUG=1788, 1667
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48669004
Cr-Commit-Position: refs/heads/master@{#8881}
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 75a025562791046c53ee5a00bee77e404a33549b |
27-Mar-2015 |
Per <perkj@chromium.org> |
Handle borked Android cameras gracefully.
It turns out that Camera.getCameraInfo can throw an exception if the camera does not work.
TESTED=added a throw before all calls to Camera.open and Camera.getCameraInfo and made sure APPRtcDemo does not crash.
BUG=4371
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44909004
Cr-Commit-Position: refs/heads/master@{#8876}
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 8ed6a4bba41ab8c707e141a210666d60dc4d170d |
27-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Remove unused non-standard capture stats.
Removes 'googCaptureJitterMs' and 'googCaptureQueueDelayMsPerS' from
talk/. The overuse-detection method used is based on encoding time,
so these stats aren't useful enough to warrant having them showing up in
GetStats().
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50469004
Cr-Commit-Position: refs/heads/master@{#8874}
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
|
| 3954e1dfe126815053c2f908214f89f9d5035a0f |
27-Mar-2015 |
Magnus Jedvert <magjed@webrtc.org> |
Remove unused implementations in cricket::VideoFrame
This CL moves dummy implementations from cricket::VideoFrame to NullVideoFrame instead.
R=guoweis@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50409004
Cr-Commit-Position: refs/heads/master@{#8873}
edia/base/nullvideoframe.h
edia/base/videoframe.cc
edia/base/videoframe.h
|
| 7100dcd3176f6522ee96be797f73a1f50da0f5d1 |
27-Mar-2015 |
Minyue Li <minyue@webrtc.org> |
Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
Specifically,
usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.
BUG=1014
R=juberti@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48499004
Cr-Commit-Position: refs/heads/master@{#8872}
pp/webrtc/webrtcsdp.cc
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 74d9ed7d853677d297807021436467a4f97584ac |
26-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Report send codec name in GetStats().
BUG=4461
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51439004
Cr-Commit-Position: refs/heads/master@{#8869}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| d6f4c25eedcfd502920f1b2a24744badd9da80be |
26-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject streams reusing simulcast or RTX SSRCs.
BUG=1788, chromium:470122, chromium:470856
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42919004
Cr-Commit-Position: refs/heads/master@{#8868}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| e59041672283a28bde0b043c0c2bc198272f82e1 |
26-Mar-2015 |
Stefan Holmer <holmer@google.com> |
Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 5225dd818047a06fe2f2a246db0fd18bb4deef5b |
26-Mar-2015 |
Brave Yao <braveyao@webrtc.org> |
If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
BUG=4289
TEST=Manual/Auto Test
R=juberti@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44629004
Cr-Commit-Position: refs/heads/master@{#8863}
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| bf395c1fc0a29b54fac4b6f6e9f6c117762faa15 |
25-Mar-2015 |
Bjorn Volcker <bjornv@webrtc.org> |
Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.
This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.
The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/
BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48699004
Cr-Commit-Position: refs/heads/master@{#8861}
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| caae5d47c1d7a479955a3445ff9e7c48de59a168 |
25-Mar-2015 |
Chuck Hays <haysc@webrtc.org> |
Bye request should use POST not GET
AppRTCDemo is failing to cleanly exit a room because it sends a GET request to /bye. The request to /bye should be a POST request. Because the /bye request is failing, the room is still marked as "full" and rejoining will fail.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47759004
Patch from Chuck Hays <haysc@webrtc.org>.
Cr-Commit-Position: refs/heads/master@{#8860}
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
|
| d4362cd3368d5fe542911c375b3a5c9f24b2f29d |
25-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject StreamParams with RTX SSRCs not in ssrcs.
BUG=1788, chromium:470122
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44859004
Cr-Commit-Position: refs/heads/master@{#8855}
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 0e209b03bf55d6daf209e35b3a8e8b6eab3d4d52 |
24-Mar-2015 |
Donald Curtis <decurtis@webrtc.org> |
Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/.
BUG=1574
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36659004
Cr-Commit-Position: refs/heads/master@{#8851}
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| e61c64dbb19b0aa58db61aecfbd9a81a253f2463 |
24-Mar-2015 |
Magnus Jedvert <magjed@google.com> |
Delete NullVideoRenderer
NullVideoRenderer is not used.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51419004
Cr-Commit-Position: refs/heads/master@{#8850}
ibjingle_tests.gyp
edia/base/nullvideorenderer.h
ession/media/channelmanager_unittest.cc
|
| 07a4ba5d1a278cd14d564408dfd06d6c9dc4e1e9 |
24-Mar-2015 |
Niklas Enbom <niklas.enbom@webrtc.org> |
Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45779004
Cr-Commit-Position: refs/heads/master@{#8849}
edia/webrtc/simulcast.cc
|
| ac27e204772c99d4df66a8c0c2c98e5ba59c1fc7 |
24-Mar-2015 |
Magnus Jedvert <magjed@google.com> |
Delete VideoAdapter::AdaptFrame
This CL deletes VideoAdapter::AdaptFrame and replaces the remaining calls with AdaptFrameResolution instead.
I do not expect this CL to fix the flaky VideoAdapterTests yet. I intend to replace FileVideoCapturer with a deterministic FakeVideoCapturer in a follow-up CL.
BUG=4317
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44769004
Cr-Commit-Position: refs/heads/master@{#8848}
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
|
| eebcab5ce99d3e8641dd92a569916b0d24e29fca |
24-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.
(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48579004
Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/sctputils.cc
pp/webrtc/statscollector.cc
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/test/mockpeerconnectionobservers.h
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/filemediaengine.cc
edia/base/filemediaengine_unittest.cc
edia/base/rtpdataengine.cc
edia/base/rtpdataengine_unittest.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvoiceengine.cc
ession/media/channel.cc
|
| e8152908281497d349da1249e41d29794f4b98e1 |
23-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Update README instructions for Android AppRTCDemo.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48679004
Cr-Commit-Position: refs/heads/master@{#8840}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8840 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/README
xamples/androidtests/README
|
| a5f6fb53ba802fcf44e3c187d798c5a53ad555df |
23-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Permit single-stream max bitrates above 2000k.
BUG=4463
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49509004
Cr-Commit-Position: refs/heads/master@{#8839}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8839 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| a197a5eed68ad29e42cde83052abdc55efbcd65f |
23-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Update libsrtp includes in preparation of roll into Chromium.
This CL is in preparation to roll the libsrtp update which landed in
https://codereview.chromium.org/936663005/ into Chromium.
BUG=https://code.google.com/p/chromium/issues/detail?id=328475
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40209004
Cr-Commit-Position: refs/heads/master@{#8838}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8838 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/externalhmac.cc
ession/media/externalhmac.h
ession/media/srtpfilter.cc
ession/media/srtpfilter_unittest.cc
|
| 39fc1d3d483b4f93801fa1f5ce71f7ba96d24a91 |
23-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Disable PeerConnectionClientTest.testLoopbackVp9
The test is flaky on Nexus 9.
BUG=4430
TBR=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44839004
Cr-Commit-Position: refs/heads/master@{#8836}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8836 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 0b44b58a3cfb6739f0d0030f151518553a0aab9f |
23-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Limit disabling of PeerConnectionEndToEndTest.Call to Windows
The test seems to be flaky only on Windows.
BUG=4464
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44829004
Cr-Commit-Position: refs/heads/master@{#8835}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8835 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| 64eb2ff0b97f360d2f5138eb43dcb7d2abb7ea43 |
23-Mar-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
iOS library build script
Script for building iOS fat libraries with armv7/arm64/x86_64.
BUG=4119
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51429004
Cr-Commit-Position: refs/heads/master@{#8834}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8834 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/build_ios_libs.sh
uild/merge_ios_libs
uild/merge_ios_libs.gyp
|
| 82e8ae4ee86e2477c3025cfaf77d6aa535b2c52d |
23-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest
The test has been flaky recently.
BUG=4464
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46689004
Cr-Commit-Position: refs/heads/master@{#8832}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8832 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| e5e92bd5565f2dfa0eafaa241e7c535bf7f92a95 |
22-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix)
In https://webrtc-codereview.appspot.com/43899004/ I managed to get some
kind of weird whitespace character in there that completely breaks Goma
and local compilation. This fixes that.
BUG=4452
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43909004
Cr-Commit-Position: refs/heads/master@{#8821}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8821 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| cfde27eeb3244518e2d3c3a2a5f0f85ddd32e3a3 |
22-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows.
The test is flaky:
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/4179
BUG=4452
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43899004
Cr-Commit-Position: refs/heads/master@{#8820}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8820 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| b789f6271a4af84f770a8957c9a3c6aab4971ed8 |
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..."
I've kicked of a roll into Chromium with out the WebRtcVideoEngine2 change, to see if it was causing the roll problems, but re-landing in the meantime.
> Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
> content_browsertests started failing around the time the change landed and rolls are failing now.
> I'm going to try rolling this back, start a roll, and then re-land.
>
> > Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
> >
> > Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> > has shown no major regressions on Chrome Canary/Dev that haven't been
> > addressed, so enabling it in time before feature freeze.
> >
> > BUG=1788
> > R=mflodman@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/44759004
>
> TBR=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/43889004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50459004
Cr-Commit-Position: refs/heads/master@{#8817}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8817 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
|
| 0c3400168acc70365beb3f1d6e78fa3707d51c65 |
22-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
content_browsertests started failing around the time the change landed and rolls are failing now.
I'm going to try rolling this back, start a roll, and then re-land.
> Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
>
> Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> has shown no major regressions on Chrome Canary/Dev that haven't been
> addressed, so enabling it in time before feature freeze.
>
> BUG=1788
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/44759004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43889004
Cr-Commit-Position: refs/heads/master@{#8816}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8816 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
|
| 4ddc9387bd064f89411a54f890f240ec17d86ed2 |
20-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Support VP8 hardware encoding and decoding on IA devices.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42829004
Cr-Commit-Position: refs/heads/master@{#8812}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8812 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| b9557a9bb7ed5f9aa1e7b3a64de4238572794ae3 |
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix code to handle crashes for non-VP8.
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.
BUG=1667, 1788
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44789004
Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 66df3cf7ab7c2fa743d428cb1b44197906810141 |
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
Removes the experiment launching WebRTC-NewVideoAPI. This field trial
has shown no major regressions on Chrome Canary/Dev that haven't been
addressed, so enabling it in time before feature freeze.
BUG=1788
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44759004
Cr-Commit-Position: refs/heads/master@{#8809}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8809 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
|
| 8296ec518b2659de922668bfe0db57e71eb17e74 |
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44779004
Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvoiceengine.h
|
| 9f9ea7e5abc3fa561e6b190b45219f2416c8786b |
20-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Clean up webrtc external capture.
This cl removes the dependency to the external capture module if external capturing is used in webrtc.
It also removes two external capture methods that is not needed.
Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input.
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43879004
Cr-Commit-Position: refs/heads/master@{#8804}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8804 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| 0c2629973929d5585d310698fab1a1e21b39d5c9 |
19-Mar-2015 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org> |
Disabling two flaky tests in libjingle_media_unittest.
BUG=4452,4453
R=kjellander@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44739004
Cr-Commit-Position: refs/heads/master@{#8791}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8791 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 8cc47e926c34d48c8b4d61146d611dd548df7f19 |
19-Mar-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Objective-C readability review.
BUG=
R=rsesek@chromium.org
Review URL: https://webrtc-codereview.appspot.com/34679004
Cr-Commit-Position: refs/heads/master@{#8784}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8784 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/objc/AppRTCDemo/ARDAppClient+Internal.h
xamples/objc/AppRTCDemo/ARDAppClient.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/tests/ARDAppClientTest.mm
|
| 840da7b75583990d414688c7c0ce1119fe5883c9 |
18-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Implement Rotation in Android Renderer.
Make use of rotation information from the frame and rotate it accordingly when we render the frame.
BUG=4145
R=glaznev@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8770
Review URL: https://webrtc-codereview.appspot.com/50369004
Cr-Commit-Position: refs/heads/master@{#8781}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8781 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 143451d2590ef951f6e66a983a38a18fcd4c66a5 |
18-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Base start bitrate on last observed bitrate.
Instead of setting bitrates based on codec target settings (which may
have previously been capped by a codec max bitrate), fetch the last
bandwidth allocated for this channel. This fixes broken low start bitrates
due to QCIF being set as default codec in WebRtcVideoEngine2 which caps
the max bitrate to 200kbps.
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43789004
Cr-Commit-Position: refs/heads/master@{#8780}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8780 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| af612d5e0769571544952cbe55e675748afa9bdd |
18-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 14ee8cc9c7c04f0125bdb1e46226918ca090a66b |
18-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoFrame: Support odd resolutions
We currently truncate the resolution of frames to a multiple of 4. This is unnecessary as everything supports odd resolutions now.
R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43819004
Cr-Commit-Position: refs/heads/master@{#8774}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8774 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer_unittest.cc
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 3fffd66dfa83494294de1b6ccd4c775f554e3be2 |
18-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Implement Rotation in Android Renderer."
This reverts commit 835ec63d8a64bbc8a573a5e0b7a09659188122d2.
TBR=guoweis@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/51399004
Cr-Commit-Position: refs/heads/master@{#8771}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8771 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 835ec63d8a64bbc8a573a5e0b7a09659188122d2 |
18-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Implement Rotation in Android Renderer.
Make use of rotation information from the frame and rotate it accordingly when we render the frame.
BUG=4145
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50369004
Cr-Commit-Position: refs/heads/master@{#8770}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8770 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 52cd828e1731266272e671020c353f5f89992a83 |
18-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Allow webrtc external encoder factories to declare encoders have internal camera sources.
This flag is passed to existing VieExternalCodec API (and others) to denote encoders that don't require/expect frames from the normal capture pipeline. This is the simplest way to allow camera->encoder texture support, until textures are supported through the normal camera pipeline and the lifetime issues are all figured out (I hear this is on the backlog, but not there yet).
Ideally, the flag would be on the encoder, but that doesn't work with SimulcastEncoderAdapter, since it doesn't create an encoder right away.
Note that this change only affects WebRtcVideoEngine (not WRVE2), since WRVE2 uses video_send_stream, and my hope is that by the time things have switched to WRVE2, textures will be supported with the normal camera pipeline and the dependency on internal sources can be thrown away.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42349004
Cr-Commit-Position: refs/heads/master@{#8769}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8769 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoencoderfactory.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 2161234cf6260feb4ec4e7e4ec1d6fd6c041df1f |
17-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add new features to AppRTCDemo from private repo.
- Add HUD fragment with HUD related controls and more
HUD statistics.
- Create and set all peer connection constraints in
PeerConnectionClient class.
- Handle registration request in web socket class internally
once web socket connection is opened.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44669004
Cr-Commit-Position: refs/heads/master@{#8762}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8762 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/layout/activity_call.xml
xamples/android/res/layout/fragment_call.xml
xamples/android/res/layout/fragment_hud.xml
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/CallFragment.java
xamples/android/src/org/appspot/apprtc/HudFragment.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/android/src/org/appspot/apprtc/util/LooperExecutor.java
|
| a78a94e838467260f80b08b83b1b5b92564d6c91 |
17-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Fix RateTracker to set an initial reference time when first updated.
BUG=4442
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43829004
Cr-Commit-Position: refs/heads/master@{#8751}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8751 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
|
| ae222b5be67e2e7df39ed282bf525e91e96c1825 |
17-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove dead code in WebRtcVideoEngine2 unittests.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43609004
Cr-Commit-Position: refs/heads/master@{#8747}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8747 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 858024f1d963c4aefc6250d68356e95095c8195f |
17-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoFrame: Initialize members in empty constructor
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41319004
Cr-Commit-Position: refs/heads/master@{#8746}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8746 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoframe.cc
|
| 592470b4ff39d60b52c745432ec131f05f3b6aa9 |
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47599004
Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ession/media/channel.cc
ession/media/channel.h
|
| 6ad507ac35ce638beddd7ac6687d006995637253 |
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE.
Also, remove channel_name. It's no longer needed.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43719004
Cr-Commit-Position: refs/heads/master@{#8741}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8741 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel.cc
ession/media/channel.h
ession/media/channelmanager_unittest.cc
|
| 4eeef584a7d6f44f65c28352762775e1d1ca8a2b |
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47589004
Cr-Commit-Position: refs/heads/master@{#8740}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8740 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ession/media/channel.cc
ession/media/channel.h
|
| c04a97f054348909c5b0c24369fb4272c2c16041 |
16-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move from BaseSession::GetStats to WebRtcSession::GetTransportStats
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
Review URL: https://webrtc-codereview.appspot.com/45639004
Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
|
| 3f11823a1a802d6073c416d32c347e7fb6b236f7 |
16-Mar-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Disables SW AEC when built-in AEC is enabled
As of r7849 the built-in AEC on devicing supporting it is enabled by default.
Unfortunately, the SW AEC (AECM) was not disabled, hence running on top of the built-in one. This is not necessary. In fact it reduce double talk performance significantly.
BUG=4431
TESTED=manually
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49419004
Cr-Commit-Position: refs/heads/master@{#8735}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8735 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| 2056ee3e3c7683ae4b2c4b12da99c3105c4f46a9 |
16-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
This reverts commit r8731.
Reason for revert: Breakes Chromium FYI bots.
TBR=hbos, tommi
Review URL: https://webrtc-codereview.appspot.com/40359004
Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediadecoder_jni.cc
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcpassthroughrender_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
|
| 93d9d6503e2bf2526af2b1c2cc46ef242b9843aa |
16-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments.
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45629004
Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediadecoder_jni.cc
|
| 2dc5fa69b2baef2ece158c9e1285516087faaa53 |
16-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*.
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40299004
Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediadecoder_jni.cc
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcpassthroughrender_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
|
| 4b89aa03bb9c817cf2274f2035d613a70c5298eb |
16-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Change StatsCollector to use DCHECK instead of ASSERT.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46579004
Cr-Commit-Position: refs/heads/master@{#8729}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8729 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
|
| eed2fcaa7631d7023c99f6480a268f9e9468f57a |
16-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision 00e438c..8d51d96 (320241:320682)
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: https://chromium.googlesource.com/chromium/src/+/00e438c..8d51d96/DEPS
This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.
Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.
Clang version was not updated in this roll.
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42779004
Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/project.properties
xamples/android/project.properties
xamples/androidtests/project.properties
|
| 2d25b44f470afdd56513b75d641166f6e7cdcd04 |
16-Mar-2015 |
changbin.shao@webrtc.org <changbin.shao@webrtc.org> |
Check associated payload type when negotiate RTX codecs.
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34189004
Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/codec.h
edia/base/rtpdataengine.cc
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| c29f7f3a5fc43018b10bba5146e3524c7466b4d6 |
14-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Disable assert for nr of threads in PeerConnectionTest.java.
This test is flaky so we need to figure out a better way to do it.
I've documented what we've observed and added a todo for myself to figure out a solution.
R=kjellander@webrtc.org
BUG=4424
Review URL: https://webrtc-codereview.appspot.com/46599004
Cr-Commit-Position: refs/heads/master@{#8725}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8725 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
|
| f1f558cde8efd4e95efd6ccb85194440a2b68a5c |
14-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix AppRTCDemo and AppRTCDemoTest builds.
On fresh checkout AppRTCDemo and corresponding tests
fail to build because resource file R.java is not auto generated properly.
On existing tree R.java will be picked up from previous
build leftover at talk/examples/android/gen.
Build bots did not detect this break for some reason.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43749004
Cr-Commit-Position: refs/heads/master@{#8723}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8723 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| d83f4eff84d872da3e38e1a61d669fc407ce7adf |
13-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8689
Committed: https://code.google.com/p/webrtc/source/detail?r=8701
Committed: https://code.google.com/p/webrtc/source/detail?r=8706
Review URL: https://webrtc-codereview.appspot.com/42659004
Cr-Commit-Position: refs/heads/master@{#8722}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8722 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory_unittest.cc
|
| b01c707209eff893223ed7af1e5fdb75b34a22a4 |
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Use a NULL session in unit tests that don't actually use the session.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49379004
Cr-Commit-Position: refs/heads/master@{#8721}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8721 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector_unittest.cc
ession/media/mediarecorder_unittest.cc
|
| b4aac13810815f77b019f9db9d0300862c8313bc |
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49399004
Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 990a00c30a2e87972506aac3a992a93ed3c8f79a |
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Remove unused transport code.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49389004
Cr-Commit-Position: refs/heads/master@{#8719}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8719 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel.h
|
| 9b2e1144df6e3622354caca00baf4a7462a0809c |
13-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Supporting Opus DTX in Voice Engine.
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.
BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43709004
Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 503a9e822a259504f6eccc4c5eac6633945e8cea |
13-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Make AppRTCDemoTest pass without Internet connection.
The AppRTCDemoTest is failing if the Android device lacks
an Internet connection (e.g. is in flight mode).
This change makes it benefit from the work done in
https://review.webrtc.org/36769004/ to work around that
limitation for loopback tests.
R=phoglund@webrtc.org
TBR=glaznev@webrtc.org
BUG=4421
TESTED=Successful run on Nexus 7 (2013) in flight mode using:
ninja -C out/Release
. build/android/envsetup.sh
adb install -r out/Release/apks/AppRTCDemo.apk
webrtc/build/android/test_runner.py instrumentation --test-apk AppRTCDemoTest --verbose --release
Review URL: https://webrtc-codereview.appspot.com/45649004
Cr-Commit-Position: refs/heads/master@{#8714}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8714 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 8372888b079b13ac765a33288def5a9d9e1387bd |
13-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns."
This reverts commit 45bc01a7172402aa4bb8d457474300533c273413.
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/47559004
Cr-Commit-Position: refs/heads/master@{#8711}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8711 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
|
| 3d3c005f36f34369509110274891c9950c59e04f |
13-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix Android peer connection client instrumentation tests.
- Updated Java VideoRenderer removes setSize() from video renderer interface.
Remove no longer valid test, which requires setSize() call before any
frame can be rendered.
- test_runner.py tries to run private member of InstrumentationTestCase class.
Workaround it by renaming private loopback test method.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47549004
Cr-Commit-Position: refs/heads/master@{#8707}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8707 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| fde1de93f96e6c624d2762214d2318653acb1853 |
13-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8689
Committed: https://code.google.com/p/webrtc/source/detail?r=8701
Review URL: https://webrtc-codereview.appspot.com/42659004
Cr-Commit-Position: refs/heads/master@{#8706}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8706 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
|
| 00c509ad1c94805b3332f2ce876c04705abf8ef5 |
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.
Tested with peerconnection_client on windows, AppRTCDemo on Mac.
BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8660
Committed: https://code.google.com/p/webrtc/source/detail?r=8661
Review URL: https://webrtc-codereview.appspot.com/43569004
Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/mediastreaminterface.h
pp/webrtc/objc/RTCVideoRendererAdapter.mm
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
xamples/peerconnection/client/linux/main_wnd.cc
xamples/peerconnection/client/main_wnd.cc
edia/base/videoframe.h
edia/base/videorenderer.h
edia/devices/carbonvideorenderer.cc
edia/devices/gdivideorenderer.cc
edia/devices/gtkvideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 04cd69887d6b81c4046adf0e7ca7f4a4909c7530 |
12-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns."
This reverts commit 93604daf0ea1fea1148ce07793bfa538d177c876.
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/40329004
Cr-Commit-Position: refs/heads/master@{#8704}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8704 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
|
| fdd10579496123c9a7fdc0bf185e2a26a12ed340 |
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add CVO support to Vie layer.
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.
WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.
BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429007
Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/fakewebrtcvideoengine.h
|
| 4f85288e71136671ae194fcdd730e2d0f0241db9 |
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Socket options are only applied when first setting TransportChannelImpl.
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.
Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.
BUG=4374
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42699004
Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
ession/media/channel.h
|
| 93604daf0ea1fea1148ce07793bfa538d177c876 |
12-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8689
Review URL: https://webrtc-codereview.appspot.com/42659004
Cr-Commit-Position: refs/heads/master@{#8701}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8701 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
|
| d3900296ae4416de2ea21be4548ea4adba8f3280 |
12-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Use a variant for storing stats values in StatsCollector code.
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports. As is, that code is intertwined, so we populate the stats on both signaling and worker thread.
I'm also adding some documentation and TODOs for further improvements.
BUG=2822
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47459004
Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 75b7f17c29565ac8ddba38c14239113ac471ca5a |
12-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Temporary interface change to StatsReport::Id.
This change is just to allow rolling into Chromium, update Chromium and then commit the actual change in WebRTC that requires the interface change. It allows using a StatsReport::Id object as a pointer (foo->Bar()), since in an upcoming change, Id objects will be pointers.
R=magjed@webrtc.org
BUG=2822
Review URL: https://webrtc-codereview.appspot.com/43689004
Cr-Commit-Position: refs/heads/master@{#8697}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8697 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.h
|
| afdd5dd372d69be7244a3d90d70de9d5ecd60eb9 |
12-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
This reverts r8683 and is a reland of r8682.
Reason for revert: The thread checker in Chromium that crashed has been fixed now.
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/40319004
Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| e9413c686ef41c051b820d5650d547caf741b56e |
12-Mar-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Revert 8689 "Fix an issue in DtlsIdentityStore when the store is..."
> Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
>
> BUG=crbug/464995
> R=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/42659004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42729004
Cr-Commit-Position: refs/heads/master@{#8690}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8690 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
|
| 2a3942adc6aeac629c56adfebaae002cdff6f186 |
12-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
BUG=crbug/464995
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42659004
Cr-Commit-Position: refs/heads/master@{#8689}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8689 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
|
| 8c5ea8a811ad4488c44b08e9621472a863b5e824 |
11-Mar-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Fix temporal layer log string.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43639004
Patch from Noah Richards <noahric@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8687}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8687 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/simulcast.cc
|
| ae1a078ac45a1f78bae72fbe5c70d37b1056b8e1 |
11-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Convert AppRTCDemo and AppRTCDemoTest to proper GYP target.
Initial CL for converting AppRTCDemo and AppRTCDemoTest to
the Chromium style of APK targets. This would
make it possible to get rid of all the ugly
bash stuff we currently have.
CL will bump minimum SDK to v14, but this is the requirement to use Chrome tools.
Initial work was done by kjellander@
https://webrtc-codereview.appspot.com/44549005/
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43679004
Cr-Commit-Position: refs/heads/master@{#8686}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8686 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/AndroidManifest.xml
xamples/android/res/layout/activity_connect.xml
xamples/android/res/layout/fragment_call.xml
xamples/android/res/values-v17/styles.xml
xamples/android/res/values/styles.xml
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
ibjingle_examples.gyp
|
| b218ff553148b9a26c82e3b3a46d626c4438cedd |
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
This reverts r8682.
Reason for revert: Fails on Chromium FYI content_browsertests
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/47529004
Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 370a72cc3ff928099c6ec6766659ed12155b74df |
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove frame copy from cricket::VideoFrame to I420VideoFrame
BUG=1128
R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42249004
Cr-Commit-Position: refs/heads/master@{#8682}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| e77c9c8df54d6a14a27e7c5e16bf55fb121426ef |
11-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Build WebRtcMediaEngine2 outside of Chromium.
Removes #ifdef WEBRTC_CHROMIUM_BUILD from
talk/media/webrtc/webrtcmediaengine.cc. WebRtcVideoEngine2 is built on
all platforms so there's no longer any need to guard this code under
ifdefs.
BUG=1788
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42719004
Cr-Commit-Position: refs/heads/master@{#8679}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8679 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
|
| 9bfa5f0405473e974792e986a6492e67cb41625d |
11-Mar-2015 |
braveyao@webrtc.org <braveyao@webrtc.org> |
In r8605, DTLS is enabled by default for native webrtc. So we have to disable it explicitly in peerconnection example for loopback test.
BUG=4386
TEST=Manual Test
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44599004
Cr-Commit-Position: refs/heads/master@{#8677}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8677 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/client/conductor.cc
|
| fc516077ed87ee99f420b3d76eb96bae3775714d |
10-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix Android AppRTCDemo failure on devices with one or no camera.
- Disable video call on devices with no camera.
- Open default camera and disable camera switch on
devices with one camera.
BUG=4373
R=braveyao@webrtc.org, wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46539004
Cr-Commit-Position: refs/heads/master@{#8674}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8674 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 4052d881620f3f79a63df3d779f93d6124a5dc63 |
10-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove GetLastRenderedFrame
This function is not used.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40269004
Cr-Commit-Position: refs/heads/master@{#8673}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8673 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcpassthroughrender.h
|
| d7452a016812ab1de69c3d7a53caca5b06c64990 |
10-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
This reverts commit r8633.
Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.
BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj
Review URL: https://webrtc-codereview.appspot.com/46549004
Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| aa57702c08e130e37e78b5ba32c816ab4f04a0c7 |
10-Mar-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
Removed texture_video_frame.h and webrtctexturevideoframe.h
BUG=1128
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45579004
Cr-Commit-Position: refs/heads/master@{#8667}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8667 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtctexturevideoframe.h
|
| f9a75d99b92402c56744121b7bc991a9c71cf324 |
10-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Add concept of whether video renderer supports rotation."
This reverts commit 0ad48935fc5b92be6e10924a9ee3b0dc39c79104.
TBR=guoweis@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/41199004
Cr-Commit-Position: refs/heads/master@{#8663}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8663 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/mediastreaminterface.h
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videorenderer.h
edia/devices/carbonvideorenderer.cc
edia/devices/gdivideorenderer.cc
edia/devices/gtkvideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 60a2aa06527d3cb7f215d2c3e6284d92af7cf6fd |
10-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Add concept of whether video renderer supports rotation."
This reverts commit 31d16467aceac56c3cb87a84564ea5e45a49ffe4.
TBR=guoweis@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/47489004
Cr-Commit-Position: refs/heads/master@{#8662}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8662 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 31d16467aceac56c3cb87a84564ea5e45a49ffe4 |
10-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.
BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8660
Review URL: https://webrtc-codereview.appspot.com/43569004
Cr-Commit-Position: refs/heads/master@{#8661}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8661 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 0ad48935fc5b92be6e10924a9ee3b0dc39c79104 |
10-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add concept of whether video renderer supports rotation.
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.
BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43569004
Cr-Commit-Position: refs/heads/master@{#8660}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8660 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/mediastreaminterface.h
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videorenderer.h
edia/devices/carbonvideorenderer.cc
edia/devices/gdivideorenderer.cc
edia/devices/gtkvideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 67186fe00cc68cbe03aa66d17fb4962458ca96d2 |
09-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix clang style warnings in webrtc/base
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.
Not inlining virtual functions with simple bodies such as
{ return false; }
strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.
BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47429004
Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
ICENSE_THIRD_PARTY
|
| 2989204130f9a4c20e3e903d38218df932d9f69d |
09-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix instability in peer connection client unit test.
- Add a separate thread to process peer connection ICE messages
to void setting remote ICe candidate in local ICE candidate callback.
- Set proper constraints values.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42279004
Cr-Commit-Position: refs/heads/master@{#8655}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8655 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 474d1eb22376898b36bcd04b0ce3860fa12fd984 |
09-Mar-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Adds C++/JNI/Java unit test for audio device module on Android.
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| dc08a230da9f6bb21299b01e4e0cbf12b11e0605 |
07-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix H.264 start code position search.
This will address incorrect start code search
in a sequence like 00 00 00 00 00 01.
Thanks Noah.
R=noahric@chromium.org, wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41159004
Cr-Commit-Position: refs/heads/master@{#8639}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8639 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediaencoder_jni.cc
|
| 1af1391b4119b4dfdfe4801714b19a676fbfd314 |
06-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove WebRtcTextureVideoFrame
WebRtcTextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of WebRtcTextureVideoFrame into the base class. Then it's possible to completely remove WebRtcTextureVideoFrame and all its files.
BUG=1128
R=pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48399004
Cr-Commit-Position: refs/heads/master@{#8638}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8638 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtctexturevideoframe_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| c2008a0e8cd34cb2ff40d76c02934b051a9a14bb |
06-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
RTCOpenGLVideoRenderer: Add support for padded frames
This CL allows RTCOpenGLVideoRenderer to handle frames with pitch > width by making an intermediate frame copy.
BUG=4381,1128
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46509004
Cr-Commit-Position: refs/heads/master@{#8637}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8637 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCOpenGLVideoRenderer.mm
|
| b4cd093f41bb9de42fedae1767444bef1d178aac |
06-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Change the unintentioal CHECK to DCHECK in DtlsIdentityStore.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41139004
Cr-Commit-Position: refs/heads/master@{#8636}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8636 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentitystore.cc
|
| a2a6fe66a39797ea61a04d80ce3afc494d850bfc |
06-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Reconfigure default streams on AddRecvStream.
Makes sure RTX can be used for streams that have received early media
before being properly configured.
BUG=1788
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46499004
Cr-Commit-Position: refs/heads/master@{#8634}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8634 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| bcead305a2f27c30c72c6a3824fdf12f4b83c2eb |
06-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 45cdcce5f5c34d9321915473d8a0daafcf3abf78 |
06-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove TextureVideoFrame
TextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of TextureVideoFrame into the base class. Then it's possible to completely remove TextureVideoFrame and all its files. Also, there is no point in having I420VideoFrame virtual anymore.
R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/40229004
Cr-Commit-Position: refs/heads/master@{#8629}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8629 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/native_handle_impl.h
|
| e41ec818a7fedd9d88dc8018b711ebcdab0afffd |
06-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove libjingle_root GYP variable
It is no longer needed.
R=andrew@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44449004
Cr-Commit-Position: refs/heads/master@{#8627}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8627 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| 818c4984e44fc74e5d226f270e6d91910ab8ad24 |
06-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Modify the simulcast encoder factory adapter to allow external encoder factories that support more than one codec.
Only VP8 encoders will be wrapped in the simulcast adapter; other codec types will be created directly with the real encoder factory and cleaned up appropriately.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40169004
Cr-Commit-Position: refs/heads/master@{#8623}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8623 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 2386d6dd92f10a715f131b5ad408b1babc1f35b0 |
05-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert 8599 "Revert 8580 "Unify underlying frame buffer in I420VideoFrame and...""
It's possible to build Chrome on Windows with this patch now.
BUG=1128
> This is unfortunately causing build problems in Chrome on Windows.
>> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
>>
>> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
>>
>> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
>>
>> Some additional minor changes are:
>> * Disallow creation of 0x0 texture frames.
>> * Remove the half-implemented ref count functions in I420VideoFrame.
>> * Remove the Alias functionality in WebRtcVideoFrame
>>
>> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
>> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
>> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
>>
>> BUG=1128
>> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
>>
>> Review URL: https://webrtc-codereview.appspot.com/42469004
R=pbos@webrtc.org
TBR=mflodman, pbos, perkj, tommi
Review URL: https://webrtc-codereview.appspot.com/45489004
Cr-Commit-Position: refs/heads/master@{#8616}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8616 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframefactory.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvideoframefactory.cc
|
| 5af41aabae428f261702ef287d8f07b198a7f9ba |
05-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Fix uninitialized variable. If FindConstraint() returns false, we check |value| in two places and at that point, it can hold an uninitialized value. Caught by Linux Memcheck builder.
http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/3351/steps/libjingle_peerconnection_unittest/logs/0A34BA777AB03D08
TBR=perkj@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/43579004
Cr-Commit-Position: refs/heads/master@{#8611}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8611 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
|
| bbce5efaa6155f31366cdd07c24197a0ae5f671e |
05-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8582
Committed: https://code.google.com/p/webrtc/source/detail?r=8607
Review URL: https://webrtc-codereview.appspot.com/43529004
Cr-Commit-Position: refs/heads/master@{#8609}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8609 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| d43b2c098d8c841ed8834eb39d7cd2c5b15e87c1 |
05-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit 86c33e3a94f51f8e4b4f305708ec327786ad3794.
TBR=guoweis@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/47409004
Cr-Commit-Position: refs/heads/master@{#8608}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8608 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 86c33e3a94f51f8e4b4f305708ec327786ad3794 |
05-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8582
Review URL: https://webrtc-codereview.appspot.com/43529004
Cr-Commit-Position: refs/heads/master@{#8607}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8607 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 61e00b0bcab899a32f14c1e2e0f4b7f316cc1f03 |
04-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8576
Committed: https://code.google.com/p/webrtc/source/detail?r=8581
Review URL: https://webrtc-codereview.appspot.com/37889004
Cr-Commit-Position: refs/heads/master@{#8605}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8605 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| f7abb12aa9919203210813eb853729a7fc2cfe07 |
04-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Fix OVERRIDE->override again after reverting video frame cl.
TBR=magjed@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/40199004
Cr-Commit-Position: refs/heads/master@{#8600}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8600 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoframe.h
|
| 1f94407319f85abc286c993774a4ea93807ec32e |
04-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8580 "Unify underlying frame buffer in I420VideoFrame and..."
This is unfortunately causing build problems in Chrome on Windows.
> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
>
> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
>
> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
>
> Some additional minor changes are:
> * Disallow creation of 0x0 texture frames.
> * Remove the half-implemented ref count functions in I420VideoFrame.
> * Remove the Alias functionality in WebRtcVideoFrame
>
> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
>
> BUG=1128
> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/42469004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42199005
Cr-Commit-Position: refs/heads/master@{#8599}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8599 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframefactory.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvideoframefactory.cc
|
| 92f4018d80ec8b092b7c1a35528e57e926f75111 |
04-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do.
Rename "AddValue" methods to AddXxx where Xxx is the type being added. Moving forward, we'll support those types natively without conversion to string.
Normalizing the extraction code to have fewer places that add the same stats and data driven additions to reports instead of multiple call sites.
BUG=2822
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47369004
Cr-Commit-Position: refs/heads/master@{#8597}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8597 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/audiotrack.h
pp/webrtc/audiotrackrenderer.h
pp/webrtc/dtmfsender.h
pp/webrtc/dtmfsender_unittest.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/mediastream.h
pp/webrtc/mediastreamhandler.h
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/RTCVideoRendererAdapter.mm
pp/webrtc/peerconnection.h
pp/webrtc/proxy.h
pp/webrtc/remoteaudiosource.h
pp/webrtc/remotevideocapturer.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/videosource.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakemediaengine.h
edia/webrtc/fakewebrtccommon.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcpassthroughrender.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
edia/webrtc/webrtcvideoframefactory.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| 058b1f17ac43b1fe69a8c18aaa7999ba88733dfd |
04-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.
BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47379004
Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| fc2f146af22dc5ba8d53d6e9ff1b7a93fa412d24 |
04-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit bbbdeed2bff31777ca7d298d17336fe94626f5b3.
TBR=juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/41109004
Cr-Commit-Position: refs/heads/master@{#8585}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8585 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 7bea1ffe772e837d96f8faa5c9dd06e531b95379 |
04-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Expose negotiated ciphers through stats API.
Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.
BUG=3976
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35169004
Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| be77872d2ce7a5faf15d3794635456ee81a5ced1 |
04-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
Breaking Chromium FYI.
TBR=pthatcher@webrtc.org
This reverts commit 369f68255ffd3d6f3e449e0defeae820cefd4f29.
BUG=4241
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8576
Review URL: https://webrtc-codereview.appspot.com/37889004
Review URL: https://webrtc-codereview.appspot.com/47389004
Cr-Commit-Position: refs/heads/master@{#8583}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8583 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| bbbdeed2bff31777ca7d298d17336fe94626f5b3 |
04-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43529004
Cr-Commit-Position: refs/heads/master@{#8582}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8582 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 369f68255ffd3d6f3e449e0defeae820cefd4f29 |
04-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8576
Review URL: https://webrtc-codereview.appspot.com/37889004
Cr-Commit-Position: refs/heads/master@{#8581}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8581 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| c8895aa2f31e05d3bd4d29507af3bbfcaa638499 |
03-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
Some additional minor changes are:
* Disallow creation of 0x0 texture frames.
* Remove the half-implemented ref count functions in I420VideoFrame.
* Remove the Alias functionality in WebRtcVideoFrame
The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
* Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
* Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42469004
Cr-Commit-Position: refs/heads/master@{#8580}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8580 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframefactory.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvideoframefactory.cc
|
| 8ad96605c1b7e77237358f4fd4c596480ee08738 |
03-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
Test failure: http://chromegw/i/client.webrtc/builders/Linux32%20Release/builds/3557
This reverts commit df512cc8b73ff519dcdf63a2603ab312d3443402.
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41089004
Cr-Commit-Position: refs/heads/master@{#8579}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8579 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| df512cc8b73ff519dcdf63a2603ab312d3443402 |
03-Mar-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Create a in-memory DTLS identity store that keeps a free identity generated in the background.
BUG=4241
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37889004
Cr-Commit-Position: refs/heads/master@{#8576}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8576 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| a1c9803e3283d41da0256779a01e355a881d407b |
03-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Fix crash in setPictureSize on Galaxy Nexus.
This cl tries to find the best supported pictureSize before setting it.
BUG=4197
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45419004
Cr-Commit-Position: refs/heads/master@{#8571}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8571 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| be00e3c198f00bdbc81ea4a00ea0893b2097f543 |
03-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Make sure VideoFrameFactory handles rotated frames when scaling.
BUG=4366
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41079004
Cr-Commit-Position: refs/heads/master@{#8570}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8570 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframefactory.cc
|
| cb04aa4a815d0b11ed6d9caa56d183cbe983cd68 |
03-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoFrameTest: Initialize memory to fix DrMemory error
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41029004
Cr-Commit-Position: refs/heads/master@{#8566}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8566 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 1d82813961d49e1a433024221b6f7164856635ec |
03-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Fix CVO in androidvideocapturer".
This cl was originally revieved in https://webrtc-codereview.appspot.com/40759004/
Patchset 2 adds a unittest for VideoFrame::Reset with and without the apply_rotation flag set.
BUG=4145
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42559004
Cr-Commit-Position: refs/heads/master@{#8564}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8564 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
edia/base/videoadapter_unittest.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 0482d0190259322aab4b8ee45a74cb621b383de2 |
02-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Implement TraceCallback in a nested class of WebRtcVideoEngine.
This is to fix a race that occurs in unit tests when the tests inherit
from the engine class that also implements the callback interface for
tracing. If tracing happens while the most derived class is still being
constructed, we're in trouble.
So, instead, factoring out the TraceCallback implementation.
R=pbos@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/43489004
Cr-Commit-Position: refs/heads/master@{#8562}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8562 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 43f4a47c2895beb0d7bf24a8cc1f3237133d99cb |
02-Mar-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add more Android peer connection client unit tests:
- Add front/back camera switch test.
- Add video source stop and restart test to simulate
application going into background.
- Add a loopback test for 3 video codecs - VP8, VP8, H.264.
- Add a loopback test for voice only call.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43459004
Cr-Commit-Position: refs/heads/master@{#8560}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8560 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| f1f0d9a4cd53f4eacbf791cb7317612fa7382a45 |
02-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEngine::SetVoiceEngine.
Instead enforcing that a voice engine is set on construction. Apart from
simplifying the class this permits tracing to be set up in the
constructor without worrying about racing sets from SetVoiceEngine
later.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44489004
Cr-Commit-Position: refs/heads/master@{#8555}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8555 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakemediaengine.h
edia/base/mediaengine.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 60f9d6f9591dba324545e976156dd27118d049f1 |
02-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Revert "Add default implementation to VideoSourceInterface."
Chrome test mock has been updated so VideoSourceInterface can now be pure virtual again. This reverts commit ed8d52378c43a7a93e0d2ca586486ca06db9eabe.
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45399004
Cr-Commit-Position: refs/heads/master@{#8551}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8551 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/videosourceinterface.h
|
| afa6d16a05301c462ff65aa4f1537a1aa12a0a7a |
02-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Add a ToString() method to StatsReport::Value.
This is an interface change only at this point which will be followed up by a matching change in Chromium that removes the dependency on the 'value' member variable. Once that's been done, I'll add native support for non-string types in the Value class.
R=magjed@webrtc.org
BUG=2822
Review URL: https://webrtc-codereview.appspot.com/40139004
Cr-Commit-Position: refs/heads/master@{#8550}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8550 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 50b229509187cf63b5c80ff5ae55694f0e84ee23 |
02-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoFrameFactory: Don't overwrite frames in use
VideoFrameFactory has a single frame buffer that is used when scaling frames. If the previous frame is still in use, we need to allocate a new frame.
BUG=4347
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36359004
Cr-Commit-Position: refs/heads/master@{#8549}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8549 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.h
edia/base/videoframefactory.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 24485eb3cc7e2729613d9fa413e476ef91977871 |
02-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove last pieces of libjingle_unittest
Most of this code has been moved into rtc_unittests
a long time ago. The target is no longer executing on the bots.
BUG=
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39319004
Cr-Commit-Position: refs/heads/master@{#8548}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8548 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 5cd6828ee6d0b271f0c54b3fae17eebd9d08c573 |
02-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove stale isolate files.
These two tests no longer exist, they're a part of
the rtc_unittests target.
The libjingle_unittest target is being completely removed in
https://webrtc-codereview.appspot.com/39319004/
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38349004
Cr-Commit-Position: refs/heads/master@{#8547}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8547 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_sound_unittest.isolate
ibjingle_unittest.isolate
|
| d68fa65d76f11f8294cd2852e2b1c3c28fc2465a |
28-Feb-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Improve cleaning for Android demo applications
There are a bunch of directories that are not cleaned between
builds since they're added to .gitignore.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40999004
Cr-Commit-Position: refs/heads/master@{#8542}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8542 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| 73acc15c69e74db7abcce7b2a27e192326bf2498 |
28-Feb-2015 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Revert 8538 "Reland "Fix CVO in androidvideocapturer."""
> Reland "Fix CVO in androidvideocapturer.""
> This reverts commit b8bcf8cbbf84971e2ae26d91659afdc58617b054.
> after I fixed a rebase mistake. The fix is the delta between patchset 1 and 2.
>
> The original cl was reviewed here:
> https://webrtc-codereview.appspot.com/40759004/
>
> TBR=magjed@webrtc.org
>
> BUG=4145
>
> Review URL: https://webrtc-codereview.appspot.com/45409004
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44439004
Cr-Commit-Position: refs/heads/master@{#8539}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8539 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
edia/base/videoadapter_unittest.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 3a93e33c56d1c88cd4ebcec272e374725065a9c1 |
27-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Fix CVO in androidvideocapturer.""
This reverts commit b8bcf8cbbf84971e2ae26d91659afdc58617b054.
after I fixed a rebase mistake. The fix is the delta between patchset 1 and 2.
The original cl was reviewed here:
https://webrtc-codereview.appspot.com/40759004/
TBR=magjed@webrtc.org
BUG=4145
Review URL: https://webrtc-codereview.appspot.com/45409004
Cr-Commit-Position: refs/heads/master@{#8538}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8538 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
edia/base/videoadapter_unittest.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| b8bcf8cbbf84971e2ae26d91659afdc58617b054 |
27-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Revert "Fix CVO in androidvideocapturer."
This reverts commit 02ed57bf9d12a959d5ec139b3fc49170d16b5f30.
https://webrtc-codereview.appspot.com/40759004/
Reason- breaks tests after rebase.
TBR=magjed@webrtc.org
BUG=4145
Review URL: https://webrtc-codereview.appspot.com/39349004
Cr-Commit-Position: refs/heads/master@{#8537}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8537 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
edia/base/videoadapter_unittest.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 02ed57bf9d12a959d5ec139b3fc49170d16b5f30 |
27-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Fix CVO in androidvideocapturer.
This add bool apply_rotation to WebrtcVideoFrame::Init and removes the need for WebrtcVideoFrame::SetRotation.
BUG=4145
R=guoweis@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40759004
Cr-Commit-Position: refs/heads/master@{#8536}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8536 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
edia/base/videoadapter_unittest.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 41d8fda12dbd1f47a4eea7e5b9995bff07bad2d8 |
27-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
VideoCapturerAndroid allocates direct buffers so that the frame buffers can be used in C++ without a copy. However byte[] array = ByteBuffer.array() seems to point to the beginning of the underlaying buffer and that is what the camera fills. But it turns out that ByteBuffer.arrayOffset() returns an offset and it seems like the pointer returned by jni->GetDirectBufferAddress(j_frame). This cl reverts back to pass the byte[] to c++ and use jni->GetByteArrayElements to get the address of the buffer.
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35349004
Cr-Commit-Position: refs/heads/master@{#8535}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8535 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 21ad37528e537a8e0f640b769c6a1804c2c7a272 |
27-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Ensure we set the right attrib for correct shader
When using oesProgram, we still specify the yuvProgram for setting shader attributes. This should be changed to the correct shader program.
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45379004
Cr-Commit-Position: refs/heads/master@{#8533}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8533 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| f296859c83c74c77e6fef7e4888b94c62661e5cf |
27-Feb-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
PeerConnectionClient.createPeerConnectionClient was calling new PeerConnectionParameters and PeerConnectionClient.createPeerConnectionFactory, .createPeerConnection with invalid arguments.
This CL makes sure the project compiles, it does not ensure the parameters now used are correct!
There may be something strange going on with the build files. I was previously able to recompile the whole project despite of the incorrect code, until I changed the file and tried again.
The changes made are just so that it will compile. The code should likely be updated by someone who knows what he/she is doing.
TBR=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45369004
Cr-Commit-Position: refs/heads/master@{#8526}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8526 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| c68e0c9dfe92fe546ab40d32660f3f1b8b5d4bf4 |
27-Feb-2015 |
braveyao@webrtc.org <braveyao@webrtc.org> |
Fix cpplint warning in the previous cl to peerconnection client example.
BUG=3872
TEST=Manual Test + AutoTest
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40949004
Cr-Commit-Position: refs/heads/master@{#8525}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8525 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
|
| ea89495786f29db5369e89a4c7ea59780e0c6787 |
27-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove {Is,Set}BlackOutput from VideoAdapter.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39309004
Cr-Commit-Position: refs/heads/master@{#8523}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8523 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/webrtc/webrtcvideoengine.cc
|
| 9650ab4d59f3a3ca6e9f738eb9dd94ff00822ca4 |
26-Feb-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Fix case sensitivity of AppRTCDemo include dirs
BUG=4341
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40939004
Patch from Vicken Simonian <vsimon@gmail.com>.
Cr-Commit-Position: refs/heads/master@{#8521}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8521 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| 2a72c6506a49b15d5e079eaa28cb80abb445684b |
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Keep feedback params in SetDefaultEncoderConfig.
Prevents NACK etc. from breaking completely as it won't be reported in
the generated SDP.
BUG=1788
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40109004
Cr-Commit-Position: refs/heads/master@{#8519}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8519 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| ac2d27d9ae74eb8d28ec0d5f12f70fa64461ab90 |
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix style violations in common_types.h and config.h
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| 891d48393e5ccd2f5e03d509c544c00a3d88cbbc |
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up target_media_bitrate in VideoSendStream.
Also wires up target_enc_bitrate in WebRtcVideoEngine2.
BUG=1667,1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42479004
Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 3e6e271ec3253e78ae0eb72156e5236d43f8731d |
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement CpuOveruseMetrics as callbacks.
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.
BUG=1667, 1788
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42429004
Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 9a4410e9934578e84cc129b978a29e151d957994 |
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement adaptation stats in WebRtcVideoEngine2.
BUG=1788
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42489004
Cr-Commit-Position: refs/heads/master@{#8510}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8510 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| b5e60b6ca7746df6fddb45e12402fa2cdd8bfe59 |
25-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Remove non necessary check from WebSocket send function.
Peer connection may generate answer and ICE candidates before
websocket client is registered. Remove check from sendAnswer()
and sendLocalIceCandidate() functions and allow websocket client
to accumulate messages and send them later once it will be
registered.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44379004
Cr-Commit-Position: refs/heads/master@{#8508}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8508 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
|
| f09e7b8a4f521447ea56e3e8c5ff2f6826feacf2 |
25-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoFrame: DCHECK exclusive ownership for non-const pixel access
Add some const safety by DCHECK(HasOneRef()) in non-const GetYPlane. This CL also replaces all incorrect non-const calls with const calls for pixel data access in cricket::VideoFrame. It's easy to call the non-const version of e.g. GetYPlane by mistake, even if only const-access is needed. For example:
const scoped_ptr<cricket::VideoFrame> foo;
const uint8_t* y = foo->GetYPlane();
will actually call the non-const version of GetYPlane.
R=mflodman@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39079004
Cr-Commit-Position: refs/heads/master@{#8507}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8507 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCI420Frame.mm
edia/base/videoadapter_unittest.cc
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe.cc
|
| 09c77b95bb62566be64da662f0b3b6a838ec6553 |
25-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add decoder-timing stats to VideoReceiveStream.
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667, 1788
Review URL: https://webrtc-codereview.appspot.com/40819004
Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 4aef5fef18011de50b2e0ebba8a99938940416f0 |
25-Feb-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
Add thread checks to the CaptureManager.
It looks like it is being used single threadedly, except that in some cases it is created and/or destroyed in threads other than the one running its operations. As such, CaptureManager() contains 'thread_checker_.DetachFromThread()' and ~CaptureManager() does not have a DCHECK.
BUG=
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36279004
Cr-Commit-Position: refs/heads/master@{#8498}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8498 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/base/capturemanager.cc
edia/base/capturemanager.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 1e64263b90022e914ca314f434b81ed4d3ce5b46 |
25-Feb-2015 |
hbos@webrtc.org <hbos@webrtc.org> |
Thread-safe ChannelManager.GetSupportedFormats, used by VideoSource
VideoSource was using VideoCapturer's GetSupportedFormats in a non-thread safe manner.
Now this is handled to (new method) ChannelManager.GetSupportedFormats.
BUG=
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42079004
Cr-Commit-Position: refs/heads/master@{#8495}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8495 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/videosource.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 112f127170193bf022565b112b03827c025168b6 |
25-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Refactor how VideoCapturerAndroid delivers frames and is stopped.
With this cl, video buffers are now allocated using direct buffers.
These buffers are guaranteed to live as long as the capturer is running.
We can now post frames in c++ from the Java thread to the c++ worker thread and let c++ post the buffers back when it has finished
processing them.
This cl also reverts back to make Stop asynchronouse so that it is guaranteed that the c++ worker thread is not used and no frames are delivered to VideoCapturerAndroid after Stop completes.
BUG=4318
TESTED= On a N5, N6, N9 and Samsung device.
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43369004
Cr-Commit-Position: refs/heads/master@{#8493}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8493 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| a4623d26d74f93591d442323bae9312eb7f07f51 |
25-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix H.264 HW decoding for Qualcomm KK devices.
- Qualcomm H.264 HW decoder on KK and older requires
a few video frames before it can generate output. Increase
maximum allowed pending frames for H.264 decoder to 30.
Plus changes in the logging to track decoder buffers
timestamps.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36319004
Cr-Commit-Position: refs/heads/master@{#8490}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8490 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 348072845a10f30257f526b658ede18490ca4e35 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Swap decl-terms from juberti@ review.
Cr-Commit-Position: refs/heads/master@{#8487}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8487 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
|
| 3630085df1042f4f42bfc5fae9dc373bac652478 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Tested equiv classes of DTLS/SCTP.
Cr-Commit-Position: refs/heads/master@{#8486}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8486 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
|
| 91d52305ac954b849352495df8500a6ae9811b23 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Renamed string and test.
Cr-Commit-Position: refs/heads/master@{#8485}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8485 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
|
| c7848b7fd1cb858b61a525b1da8b84159b19d3d3 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Added a separate DTLS/SCTP test.
Cr-Commit-Position: refs/heads/master@{#8484}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8484 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
|
| a74709333482783cb06405626caf9555e407eba2 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
After another round of reviews.
Cr-Commit-Position: refs/heads/master@{#8483}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8483 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
ession/media/mediasession.cc
ession/media/mediasession.h
|
| 9616196c38149e9a920d59da3019f47d1d61ff85 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Merging definitions of IsSctp.
Cr-Commit-Position: refs/heads/master@{#8482}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8482 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| 12aa8a68f95eb68a006fe112fabd149fab262c56 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Post-rebase.
Cr-Commit-Position: refs/heads/master@{#8481}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8481 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| 17308695963539ed3a125ba87635b81e12fac081 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Added raw SCTP to IsSctp.
Cr-Commit-Position: refs/heads/master@{#8480}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8480 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| 871b1c373ab2170056ac792bc228ba0e3c3b38b4 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Review comments -- added IsSctp()
Cr-Commit-Position: refs/heads/master@{#8479}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8479 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| d7b6165483e7c67831bf7c161168b80be21ec3be |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Made DTLS/SCTP equivalent to UDP/DTLS/SCTP when comparing session descs in tests.
Cr-Commit-Position: refs/heads/master@{#8478}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8478 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
|
| ec97c6516f0f2540a8e040d08de86709be6ab5b4 |
24-Feb-2015 |
lally@webrtc.org <lally@webrtc.org> |
Attempt on read-only acceptance of -12.
Cr-Commit-Position: refs/heads/master@{#8477}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8477 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
|
| a30f007e457997cc7add6f791c5a9562cb70c58e |
24-Feb-2015 |
phoglund@webrtc.org <phoglund@webrtc.org> |
Fixing incorrect memset in mock class.
I got a linker warning, and I could see the memset was clearly
incorrect since the arugment order should be ptr, value, size_t.
BUG=None
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35269004
Cr-Commit-Position: refs/heads/master@{#8473}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8473 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/mockpeerconnectionobservers.h
|
| a5de951b37d57de7d7700323f4ddffa00fcae861 |
24-Feb-2015 |
phoglund@webrtc.org <phoglund@webrtc.org> |
Make Options public and not package access in pc factory.
I realized I had accidentally made the Options struct package private,
which means no client can actually use it.
BUG=4181
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35279004
Cr-Commit-Position: refs/heads/master@{#8472}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8472 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
|
| e3fccd4268d8e46c737f27a431c1dd263f312395 |
24-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Merge changes from internal repo to AppRTCDemo.
- Add a setting option to disable outgoing video in a call.
- Add an option to select audio codec.
- Add an option to specify audio bitrate for Opus codec.
- Plus add an option to select H.264 as default video codec.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42449004
Cr-Commit-Position: refs/heads/master@{#8468}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8468 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
|
| d324546ced76d4e792338af4f7d02a5cd8819f92 |
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
edia/base/codec.cc
edia/base/constants.cc
edia/base/constants.h
edia/base/rtpdump_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
|
| 722739108a9a1b30cbcb8285ce0b76762b356fb3 |
23-Feb-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
Includes GN changes from
https://webrtc-codereview.appspot.com/39249004/
Android changes for JNI were required due to
https://codereview.chromium.org/843103003
Other relevant changes:
* src/buildtools: 5c5e924..93b3d0a
* src/third_party/boringssl/src: d306f16..b180ee9
* src/third_party/icu: 4e3266f..2081ee6
* src/third_party/libvpx: 5cdd302..33bbffe
* src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7
* src/tools/gyp: 4d7c139..3464008
* src/tools/swarming_client: bdad118..1b7bfec
Details: https://chromium.googlesource.com/chromium/src/+/b0c3ed3..2c3ffb2/DEPS
Clang version was not updated in this roll.
R=dpranke@chromium.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40079004
Cr-Commit-Position: refs/heads/master@{#8466}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| b28474c7a0356f21b374f43a51602ed10f143bf4 |
23-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add H.264 HW encoder and decoder support for Android.
- Allow to configure MediaCodec Java wrapper to use VP8
and H.264 codec.
- Save H.264 config frames with SPS and PPS NALUs and append them to every key frame.
- Correctly handle the case when one encoded frame may generate several output NALUs.
- Add code to find H.264 start codes.
- Add a flag (non configurable yet) to use H.264 in AppRTCDemo.
- Improve MediaCodec logging.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43379004
Cr-Commit-Position: refs/heads/master@{#8465}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8465 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediadecoder_jni.h
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 77e11bbe834e3b096db57278d2ad7c76d8c26d66 |
23-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up preferred/nominal_bitrate to stats.
Also adds a test that shows that actual_enc_bitrate was not summed
correctly plus fixing it.
Additionally reducing locking when grabbing stats.
BUG=1778
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34319004
Cr-Commit-Position: refs/heads/master@{#8464}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8464 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 962c62475e31ccb5b1315bf646138652e273d0f5 |
23-Feb-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Refactoring WebRTC Java/JNI audio track in C++ and Java.
This CL is part II in a major refactoring effort. See https://webrtc-codereview.appspot.com/33969004 for part I.
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioTrack (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Simplified the delay estimate
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39169004
Cr-Commit-Position: refs/heads/master@{#8460}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8460 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 2ad3bb17a7e0e83ae802ef62933325bce8041966 |
23-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland patch for Switch default color format to YV12 on Android.
The new since the previous patch is that we ignore all resolutions with width % 16 != 0
since they are not tightly packed.
http://developer.android.com/reference/android/graphics/ImageFormat.html#YV12
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36269004
Cr-Commit-Position: refs/heads/master@{#8459}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8459 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 3c4668e27d14358a192a179e7696cfc3c96d6ad9 |
20-Feb-2015 |
torbjorng@webrtc.org <torbjorng@webrtc.org> |
Amend CpuMonitor fix.
Merged CpuMonitor changes.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42029005
Cr-Commit-Position: refs/heads/master@{#8445}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8445 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| f906e55de1ef4cc3078b29b680f5b0da91a2858b |
20-Feb-2015 |
torbjorng@webrtc.org <torbjorng@webrtc.org> |
Add CpuMonitor to Android ApprtcDemo
R=magjed@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38169004
Cr-Commit-Position: refs/heads/master@{#8444}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8444 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/CallFragment.java
xamples/android/src/org/appspot/apprtc/CpuMonitor.java
|
| ec45e3b290621f8b58fa0de796da6d9b049a9822 |
20-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix test race in GetStatsMultipleSendStreams.
Test now waits for stats to be filled instead of failing instantly if
they haven't been updated.
BUG=2409
R=asapersson@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36239004
Cr-Commit-Position: refs/heads/master@{#8441}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8441 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
|
| 804eb468066bc930cf862652868481740dfaad95 |
20-Feb-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Change default from GICE to ICE5245 for SDP offers
BUG=4299
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34289004
Cr-Commit-Position: refs/heads/master@{#8440}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8440 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
|
| cce874b8d2f448a800317ae375b77a7935336564 |
19-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Fix libjingle_media_unittest codec comparison issue
Missing one comparison of AudioCodec
TBR=juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/42409005
Cr-Commit-Position: refs/heads/master@{#8437}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8437 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/codec.cc
|
| bc6961fe323bf60ee9fa5f6b6569f0f64a80276d |
19-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Make webrtc 50 KB smaller by not inlining Codec.
The Codec class is a big class and objects of the Codec class are passed
around by value. That means that inlined operations would be duplicated
at many places, in particular inside STL.
By not inlining Codec methods, webrtc shrinks by 50 KB in
a Linux x64 clang build.
Total change: -54147 bytes
==========================
+2810 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/codec.cc - (gained 2920, lost 110)
-1003 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/codec.h - (gained 0, lost 1003)
-1129 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc - (gained 1660, lost 2789)
-1190 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/base/rtpdataengine.cc - (gained 1408, lost 2598)
-1747 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/session/media/mediasession.cc - (gained 803, lost 2550)
-2141 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc - (gained 1679, lost 3820)
-2250 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc - (gained 1224, lost 3474)
-2927 - Source: /usr/include/c++/4.8/bits/stl_vector.h - (gained 0, lost 2927)
-3729 - Source: /home/bratell/src/chromium/src/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine2.cc - (gained 10925, lost 14654)
-6369 - Source: /usr/include/c++/4.8/bits/vector.tcc - (gained 0, lost 6369)
-10582 - Source: /usr/include/c++/4.8/bits/stl_heap.h - (gained 0, lost 10582)
-19324 - Source: /usr/include/c++/4.8/bits/stl_algo.h - (gained 743, lost 20067)
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40729005
Cr-Commit-Position: refs/heads/master@{#8436}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8436 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/codec.cc
edia/base/codec.h
|
| e07710cc91b3dccd7fea7df3d99c304f419babda |
19-Feb-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Make SendCodec() lock-free.
Fetching the current codec for sake of gathering stats, is frequently blocked since it's done by acquiring the same lock as is held while encoding frames. This can mean tens of milliseconds.
To improve this, I'm taking advantage of the fact that the codec information is set on the same thread as is used to query the information. This means that locking isn't needed for querying this information. I'm adding checks to make sure debug builds will crash if this isn't followed.
An alternative to this approach could be to add one more lock that is specifically used for the codec information variable. This would also decouple querying codec information from the encoder itself, but still requires a lock.
This patch depends on making ThreadChecker part of rtc_base_approved:
https://webrtc-codereview.appspot.com/40539004/
BUG=2822
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37779004
Cr-Commit-Position: refs/heads/master@{#8435}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8435 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine.cc
|
| 1ed6224eafc7816f25d1906e4d709afdf2ad8f0f |
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info."
This breaks compilation outside this codebase that needs to have it
removed before.
BUG=4322
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42009004
Cr-Commit-Position: refs/heads/master@{#8432}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8432 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| 8ad05b76281e73f92051125aee81d85227c6a9bc |
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove dead stats from Video{Sender,Receiver}Info.
These stats are neither filled nor plumbed further and might as well be
removed (as proven by how easy they were to remove).
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39219004
Cr-Commit-Position: refs/heads/master@{#8430}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8430 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| 1d0fa5d352fe12092201fade249905c7e1ff974b |
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 3db042e2f09f1df7d8b5d40f30766f780848ecd9 |
19-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Stop AndroidVideoCapturer asynchronously.
The purpose is to avoid a deadlock between the C++ thread calling Stop and the Java thread that provides video frames.
BUG=4318
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35249004
Cr-Commit-Position: refs/heads/master@{#8425}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8425 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 254840692ef450d94d6a4b075eb139bb34305ec0 |
19-Feb-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Add empty files to implement a in-memory DTLS identity store without breaking Chromium build.
BUG=4241
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36199004
Cr-Commit-Position: refs/heads/master@{#8424}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8424 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtlsidentityservice.cc
pp/webrtc/dtlsidentityservice.h
pp/webrtc/dtlsidentitystore.cc
pp/webrtc/dtlsidentitystore.h
pp/webrtc/dtlsidentitystore_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
|
| 652bc37a07f5ab2559fd217c22be391b45af5b53 |
19-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding two new stats to StatsReport.
A follow up of r8415. This is to post the data to the StatsReport.
BUG=3867
TEST=chromium + netem + apprtc + chrome://webrtc-internals
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38139004
Cr-Commit-Position: refs/heads/master@{#8423}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8423 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| a744a28b92bac9a98816bc0cae0104c2ecdd0edb |
18-Feb-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Templatize and clean up codec wildcards.
BUG=4123
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39209004
Cr-Commit-Position: refs/heads/master@{#8422}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8422 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 18c92472dfb12315eb01fb83e01ceebe58e200e6 |
18-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Move Android MediaCodec encoder and decoder factories to separate files.
Move Android media encoder and media decoder factories from
peerconnection_jni.cc to androidmediaencoder_jni.cc and
androidmediadecoder_jni.cc
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36139004
Cr-Commit-Position: refs/heads/master@{#8417}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8417 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidmediacodeccommon.h
pp/webrtc/java/jni/androidmediadecoder_jni.cc
pp/webrtc/java/jni/androidmediadecoder_jni.h
pp/webrtc/java/jni/androidmediaencoder_jni.cc
pp/webrtc/java/jni/androidmediaencoder_jni.h
pp/webrtc/java/jni/native_handle_impl.h
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
|
| c0bd7be0df67735d63f5cdd302a3b85f88239874 |
18-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding two new stats to VoiceReceiverInfo
There have been requests of two new stats namely
speech_expand_rate and secondary_decoded_rate.
BUG=3867
R=henrik.lundin@webrtc.org, henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40789004
Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 8fbdcfd73f22a76747cc33aa12c46b8240948258 |
18-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Revert "Switch default color format to YV12."
This reverts commit 1c3e728aa9b886fd3ee008a5aed956759bc3f82d.
Reason: Fails test running on Nexus 9 bots - org.webrtc.VideoCapturerAndroidTest#testStartStopWithDifferentResolutions.
Note that all other tests pass so it seems like there is resolution supported by the device that can't use YV12.
TBR=glaznev@webrtc.org
BUG=4011
Review URL: https://webrtc-codereview.appspot.com/42389004
Cr-Commit-Position: refs/heads/master@{#8414}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8414 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| 1c3e728aa9b886fd3ee008a5aed956759bc3f82d |
18-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Switch default color format to YV12.
Currently N21 is used per default. But according to
http://developer.android.com/reference/android/graphics/ImageFormat.html#YV12
YV12 has been mandatory to support since api level 12.
Since YV12 and I420 is the same except for the order of planes, this format is cheaper to use.
Tested on N5, N6 and a Samsung device.
BUG=4011
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40749004
Cr-Commit-Position: refs/heads/master@{#8411}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8411 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidvideocapturer.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
|
| f68e186de317abf2fd17e55a5e3cb417a0e50e1f |
18-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove EnableMirroring and MirrorRenderStream
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35239004
Cr-Commit-Position: refs/heads/master@{#8409}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8409 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcpassthroughrender.h
|
| b4987bfc24e1e755a6c54053d09a58d1e72228bb |
18-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Send black frame with previous size when muting.
Instead of sending a black frame that's the size of the VideoFormat send
a black frame in the format we're already sending. This prevents
expensive encoder reconfiguration when the sending format is a different
resolution. This speeds up setting a null capturer (removing the
capturer) significantly as it doesn't entail an encoder reconfiguration.
R=mflodman@webrtc.org, pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/39179004
Cr-Commit-Position: refs/heads/master@{#8405}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8405 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 3864363e2c3043bd23081abe32ad13dcb6d718ed |
18-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoFrame: Refactor CopyToBuffer into base class
It’s possible to implement cricket::VideoFrame::CopyToBuffer using the virtual interface. This removes the need for subclasses to implement their own versions. This CL also fixes a bug in cricket::VideoFrame::CopyToPlanes which currently assumes that GetUPitch() == GetVPitch(), otherwise it may segfault.
I think this CL should land regardless, but the main purpose is to pave the way for for planned changes to I420VideoFrame. See https://review.webrtc.org/38879004.
R=fbarchard@google.com, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39889004
Cr-Commit-Position: refs/heads/master@{#8403}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8403 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| dd4a8da68ada4c91653271462b21a23b0319ef66 |
18-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove DISABLE_YUV flag
R=fbarchard@google.com, pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41979004
Cr-Commit-Position: refs/heads/master@{#8402}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8402 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/cpuid.cc
edia/base/testutils.h
edia/base/videocapturer.cc
edia/base/videoframe.cc
|
| bfa3c7253fc29a6c64115c49457cd69cec05932b |
17-Feb-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Don't call g_thread_init on glib >=2.31.0
g_thread_init() is deprecated in glib 2.31.0 and later. This will call
g_thread_ini() only when compiling against older versions of glib.
BUG=1971,chromium:253566
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40019004
Cr-Commit-Position: refs/heads/master@{#8400}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8400 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/client/linux/main.cc
edia/devices/gtkvideorenderer.cc
|
| e9facf8bb32a1688f2156009c755caa2904e1ac9 |
17-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Add range checks in a variety of places where the values will subsequently be
expected to be 0-127.
BUG=none
TEST=none
R=juberti@webrtc.org
TBR=henrika
Review URL: https://webrtc-codereview.appspot.com/37759004
Cr-Commit-Position: refs/heads/master@{#8399}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8399 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
edia/base/rtputils.cc
edia/base/rtputils.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| 640313ce4f3001411b42f6eae37294ebb6a6e7be |
17-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoCapturer: Remove dead code |OnIncomingCapturedEncodedFrame|
The end goal except cleanup is to remove webrtc::VideoFrame.
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36079004
Cr-Commit-Position: refs/heads/master@{#8393}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8393 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideocapturer.h
|
| 1a38a511196c0aba224467d9714d9b4504cc0538 |
17-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Add default implementation to VideoSourceInterface of Stop and Restart.
This is to make sure Chrome does not break when rolling. This should be reverted once
Chrome has been updated.
Please see:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16556/steps/compile/logs/stdio
BUG=4303
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35229004
Cr-Commit-Position: refs/heads/master@{#8391}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8391 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/videosourceinterface.h
|
| 8f605e89113ccdd02a5d68edf8e7a048ab0fdaff |
17-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Add VideoSource::Stop and Restart methods.
The purpose is to make sure that start and stop is called on the correct thread on Android. It also cleans up the Java VideoSource implementation.
BUG=4303
R=glaznev@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39989004
Cr-Commit-Position: refs/heads/master@{#8389}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8389 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
pp/webrtc/java/src/org/webrtc/VideoSource.java
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/videosource.h
pp/webrtc/videosource_unittest.cc
pp/webrtc/videosourceinterface.h
pp/webrtc/videosourceproxy.h
|
| f9b5c1b3d009887df02505d12ece2f80b2a90d44 |
17-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Removing CELT.
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.
BUG=
R=pbos@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36099004
Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager_unittest.cc
|
| 86196c4f481d7f515e54806988f763169e8b9206 |
16-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Setup encoders inexpensively before first frame.
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.
Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.
Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42369004
Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 3341b401cce2b2e8dbec55bdae4261cf0fc19012 |
13-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS.
BUG=none
TEST=none
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34029004
Cr-Commit-Position: refs/heads/master@{#8369}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8369 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| 5a7dc39277999cbfa0da053da5eacc7fee5cd307 |
13-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
This is a code clean up. No functional change intended.
Consolidate the enum for capturer/frame rotation we use through out the code base.
BUG=4145
R=mflodman@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39859004
Cr-Commit-Position: refs/heads/master@{#8365}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8365 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcdeviceinfo.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/fakewebrtcvideoengine.h
|
| 96e4db9bea49cf096044c89c94778bff525362ba |
13-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Split peerconnection_jni.cc into separate files.
For now:
java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day.
classreferenceholder - app/webrtc specific Java class loader.
androidvideocapturer_jni - the jni part of the video capturer I added.
peerconnection_jni - all the rest.
This also move all jni specifics into ns webrtc_jni to avoid naming collision.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38099004
Cr-Commit-Position: refs/heads/master@{#8363}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8363 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/androidvideocapturer_jni.cc
pp/webrtc/java/jni/androidvideocapturer_jni.h
pp/webrtc/java/jni/classreferenceholder.cc
pp/webrtc/java/jni/classreferenceholder.h
pp/webrtc/java/jni/jni_helpers.cc
pp/webrtc/java/jni/jni_helpers.h
pp/webrtc/java/jni/peerconnection_jni.cc
ibjingle.gyp
|
| 40fdb8ab9669ee22b2723d154afeeebd44a08b5d |
13-Feb-2015 |
solenberg@webrtc.org <solenberg@webrtc.org> |
Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway.
BUG=3871
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41879004
Cr-Commit-Position: refs/heads/master@{#8359}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8359 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 40367f984b2922fbfcf58d7e485ac0ef59149768 |
13-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove default video encoders for new video API.
Reduces stream creation time significantly. As a side effect also
removes default encoders for receive-only channels.
BUG=1788,1667
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37049004
Cr-Commit-Position: refs/heads/master@{#8356}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8356 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| 94eb9a6005867c09df808091d60d1e5e82958359 |
13-Feb-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Whitespace change to test gsubtreed.
BUG=chromium:438149
Cr-Commit-Position: refs/heads/master@{#8355}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8355 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/whitespace.txt
|
| e388c19a9f86541f2fedd0a17c832a4e24391fe3 |
13-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix start bitrate settings for VP9 codec in AppRTCDemo.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35169005
Cr-Commit-Position: refs/heads/master@{#8354}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8354 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| aafbec15f9e71f103587d1379ff12059d5285c48 |
12-Feb-2015 |
solenberg@webrtc.org <solenberg@webrtc.org> |
Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default.
BUG=3735
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39919005
Cr-Commit-Position: refs/heads/master@{#8351}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8351 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| 503c33666ff7c382b540296755793ddab8d4b909 |
12-Feb-2015 |
solenberg@webrtc.org <solenberg@webrtc.org> |
Re-enabling LocalP2PTestAnswerVideo and LocalP2PTestAnswerAudio test cases in peerconnection_unittest.
BUG=2288
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39919004
Cr-Commit-Position: refs/heads/master@{#8350}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8350 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| ff689be3c0c59c1be29aaa0697aa0f762566d6c6 |
12-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35079004
Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
edia/base/filemediaengine.cc
edia/base/testutils.cc
edia/base/videoadapter.cc
edia/base/videocapturer.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
|
| 006521d5bdc29677c710b53c31b937cdc0bb4941 |
12-Feb-2015 |
phoglund@webrtc.org <phoglund@webrtc.org> |
Makes libjingle_peerconnection_android_unittest run on networkless devices.
PeerConnectionTest.java currently works, but only on a device with
network interfaces up. This is not a problem for desktop, but it is a
problem when running on Android devices since the devices in the lab
generally don't have network (due to the chaotic radio environment in
the device labs, devices are simply kept in flight mode).
The test does work if one modifies this line in the file
webrtc/base/network.cc:
bool ignored = ((cursor->ifa_flags & IFF_LOOPBACK) ||
IsIgnoredNetwork(*network));
If we remove the IFF_LOOPBACK clause, the test starts working on
an Android device in flight mode. This is nice - we're running the
call and packets interact with the OS network stack, which is good
for this end-to-end test. We can't just remove the clause though since
having loopback is undesirable for everyone except the test (right)?
so we need to make this behavior configurable.
This CL takes a stab at a complete solution where we pass a boolean
all the way through the Java PeerConnectionFactory down to the
BasicNetworkManager. This comes as a heavy price in interface
changes though. It's pretty out of proportion, but fundamentally we
need some way of telling the network manager that it is on Android
and in test mode. Passing the boolean all the way through is one way.
Another way might be to put the loopback filter behind an ifdef and
link a custom libjingle_peerconnection.so with the test. That is hacky
but doesn't pollute the interfaces. Not sure how to solve that in GYP
but it could mean some duplication between the production and
test .so files.
It would have been perfect to use flags here, but then we need to
hook up gflags parsing to some main() somewhere to make sure the
flag gets parsed, and make sure to pass that flag in our tests.
I'm not sure how that can be done.
Making the loopback filtering conditional is exactly how we solved the
equivalent problem in content_browsertests in Chrome, and it worked
great.
That's all I could think of.
BUG=4181
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36769004
Cr-Commit-Position: refs/heads/master@{#8344}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8344 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/portallocatorfactory.cc
pp/webrtc/portallocatorfactory.h
pp/webrtc/statscollector.cc
|
| 1226e926e6104322d9b99026b98f515cb4d40fd4 |
11-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
split from https://webrtc-codereview.appspot.com/37029004/
This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004
BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8337
Committed: https://code.google.com/p/webrtc/source/detail?r=8338
Review URL: https://webrtc-codereview.appspot.com/39799004
Cr-Commit-Position: refs/heads/master@{#8339}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8339 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideocapturer.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoframe.h
edia/base/videoframefactory.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframefactory.cc
|
| dc7b02277cc1666dfc13b636c2ecfe53b12c9d2a |
11-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
split from https://webrtc-codereview.appspot.com/37029004/
This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004
BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8337
Review URL: https://webrtc-codereview.appspot.com/39799004
Cr-Commit-Position: refs/heads/master@{#8338}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8338 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideocapturer.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoframe.h
edia/base/videoframefactory.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframefactory.cc
|
| 20e8f227664a6747cea11e1fc1de4c018ebcc8e9 |
11-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
split from https://webrtc-codereview.appspot.com/37029004/
This is based on clean up code change at https://webrtc-codereview.appspot.com/37129004
BUG=4145
R=perkj@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39799004
Cr-Commit-Position: refs/heads/master@{#8337}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8337 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideocapturer.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoframe.h
edia/base/videoframefactory.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframefactory.cc
|
| 11426dc719f24ece3246fd8fb24ae073c49b42ed |
11-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Don't rely on webrtc/base/scoped_ptr.h to include stuff for you
webrtc/base/scoped_ptr.h doesn't need to include webrtc/base/common.h
anymore, but a couple of its users were relying on it to pull in other
things for them. Fix that, and remove the now really unnecessary
webrtc/base/common.h include.
R=andrew@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37169004
Cr-Commit-Position: refs/heads/master@{#8333}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8333 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/mobiledevicemanager.cc
|
| 83bc721c7e1760ce7f96eed11a5351fa3154f523 |
11-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Add Android specific VideoCapturer.
The Java implementation of VideoCapturer is losely based on the the work in webrtc/modules/videocapturer.
The capturer is now started asyncronously.
The capturer supports easy camera switching.
BUG=
R=henrika@webrtc.org, magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30849004
Cr-Commit-Position: refs/heads/master@{#8329}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8329 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
pp/webrtc/androidvideocapturer.cc
pp/webrtc/androidvideocapturer.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoCapturer.java
pp/webrtc/java/src/org/webrtc/VideoCapturerAndroid.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
ibjingle.gyp
|
| 7cc92aaf3767ab459cf8a42e5eef50ad555e3e90 |
11-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Use WebRtcVideoRenderFrame for texture frames.
Removes buffer/texture path separation inside WebRtcVideoEngine and
DeliverTextureFrame(). This unifies frame delivery with
WebRtcVideoEngine2 which is expected to automagically work with texture
frames after this change.
BUG=1788
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38069005
Cr-Commit-Position: refs/heads/master@{#8326}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8326 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
|
| 62f6e756730325ee7b20cf5f81e82b0a70283a05 |
11-Feb-2015 |
henrika@webrtc.org <henrika@webrtc.org> |
Refactoring WebRTC Java/JNI audio recording in C++ and Java.
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33969004
Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
edia/webrtc/webrtcvoiceengine.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| f58fe0ab2bb536dd22f30ee9aef69b0e300c38f8 |
11-Feb-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Rename GYP and GN targets for video capture+render.
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.
Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)
GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture
video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render
BUG=456815
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35099004
Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| bc357036942e361212e6f979ab258be89bc886e6 |
11-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add a method to remove an existing renderer from the internal list of Android renderers.
BUG=4290
R=jiayl@webrtc.org, mquiros@google.com
Review URL: https://webrtc-codereview.appspot.com/36089004
Cr-Commit-Position: refs/heads/master@{#8320}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8320 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| bc40324d9c673f5ba4df78590d928be3b9c62418 |
11-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Merge fixes and changed for Android AppRTCDemo from internal repo.
- Rename AppRTCDemoActivity to CallActivity and move UI controls
to a fragment.
- Add option to enable/disable statistics.
- Move peer connection and video constraints from URL to peer
connection client.
- Variable renaming.
R=jiayl@webrtc.org, wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33299004
Cr-Commit-Position: refs/heads/master@{#8319}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8319 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/AndroidManifest.xml
xamples/android/res/layout/activity_call.xml
xamples/android/res/layout/activity_fullscreen.xml
xamples/android/res/layout/fragment_call.xml
xamples/android/res/layout/fragment_menubar.xml
xamples/android/res/values/strings.xml
xamples/android/res/values/styles.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/CallActivity.java
xamples/android/src/org/appspot/apprtc/CallFragment.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
ibjingle_examples.gyp
|
| f4c10d24dc8f1c3ce6859644077d7df6fb678dcd |
10-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Always use DeliverI420Frame in WebRtcVideoEngine.
Moves native_handle() path to DeliverI420Frame and CHECKs that
DeliverFrame is not being used anymore.
R=magjed@webrtc.org, mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/38019004
Cr-Commit-Position: refs/heads/master@{#8312}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8312 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 44ae4c8b07cdf06d20d5042326b90ec9b466b664 |
10-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Support using VP9 video codec in AppRTCDemo.
- Add peer connection Java API to initialize field trial string.
- Add setting option to select VP8 or Vp9 as default video codec.
- Minor code clean up and allowing 720p portrait encoding.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39899004
Cr-Commit-Position: refs/heads/master@{#8303}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8303 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 0d852d5c27a759fe7aadc500bd7b3cadfae3deb8 |
09-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Use VideoReceiveStream as an ExternalRenderer.
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.
Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/34169004
Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 53d9012faf32eb711681fdeb31b9d0d2f9e9481b |
09-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Clean kForever from basictypes and move it to the interfaces that actually have it.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33269004
Cr-Commit-Position: refs/heads/master@{#8296}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8296 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/proxy.h
edia/devices/devicemanager.cc
edia/devices/filevideocapturer.cc
edia/devices/filevideocapturer.h
edia/devices/filevideocapturer_unittest.cc
edia/sctp/sctpdataengine_unittest.cc
|
| 8cf9bdb3fad92fd783b32152e912859d8b399c97 |
09-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove USE_WEBRTC_DEV_BRANCH.
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.
R=bjornv@webrtc.org, juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/39849004
Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
uild/common.gypi
edia/webrtc/fakewebrtccommon.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 6c930c71831b6eda6e85903c505459569a02ad9a |
09-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Cleanup: unify rotation to be enum based instead of int for degree.
Split from https://webrtc-codereview.appspot.com/37029004/
BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8257
Committed: https://code.google.com/p/webrtc/source/detail?r=8276
Committed: https://code.google.com/p/webrtc/source/detail?r=8277
Review URL: https://webrtc-codereview.appspot.com/37129004
Cr-Commit-Position: refs/heads/master@{#8288}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8288 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/nullvideoframe.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 0c7ec770ff5be89a875c989f2e99d0c24d0152a7 |
06-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Cleanup: unify rotation to be enum based instead of int for degree.
Split from https://webrtc-codereview.appspot.com/37029004/
BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8257
Committed: https://code.google.com/p/webrtc/source/detail?r=8276
Review URL: https://webrtc-codereview.appspot.com/37129004
Cr-Commit-Position: refs/heads/master@{#8277}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8277 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/nullvideoframe.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 110443aaac0f71d4fe2153648544038f3a8c404d |
06-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Cleanup: unify rotation to be enum based instead of int for degree.
Split from https://webrtc-codereview.appspot.com/37029004/
BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8257
Review URL: https://webrtc-codereview.appspot.com/37129004
Cr-Commit-Position: refs/heads/master@{#8276}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8276 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/nullvideoframe.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 9baa9ca399592b8a603fb1ac98d528c476638cf9 |
06-Feb-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Add libjingle_peerconnection_so.so to Java test dependencies.
This fix a problem where the Java test is not dependent on the so file.
BUG=4275
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33239004
Cr-Commit-Position: refs/heads/master@{#8270}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8270 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 4b320cf2149b317c9ab08fe7c7017f5756651e69 |
06-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Cleanup: unify rotation to be enum based instead of int for degree."
Reason for revert:
Compile error on bots - A subclass of cricket::VideoFrame still uses old GetRotation return type.
BUG=4145
TBR=guoweis,stefan,pthatcher
This reverts commit 3e733a43f5a1a2c170e1064d0ee0af38d710a64a.
Review URL: https://webrtc-codereview.appspot.com/34159004
Cr-Commit-Position: refs/heads/master@{#8265}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8265 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 57ac2c84dd552dd7a56c5643163b1a5ce1dbf2ba |
06-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Default destination used by c line should be IPv4 only to avoid parsing error in legacy client.
Make sure the IP family overwrites the preference of candidates. Also,
make sure only UDP is used as default destination.
BUG=4269
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36009004
Cr-Commit-Position: refs/heads/master@{#8258}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8258 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 3e733a43f5a1a2c170e1064d0ee0af38d710a64a |
06-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Cleanup: unify rotation to be enum based instead of int for degree.
Split from https://webrtc-codereview.appspot.com/37029004/
BUG=4145
R=pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37129004
Cr-Commit-Position: refs/heads/master@{#8257}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8257 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| f6932297e7ac122cd3e372868ad17ccbcb8b521a |
05-Feb-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix Android video renderer to support video frames
with stride > width.
Recent libvpx update generates output video frames with stride
value greater than width, which was not supported by Android OpenGL
video renderer (Android GLES2 doesn't have GL_UNPACK_ROW_LENGTH
to provide stride information for buffer in glTexImage2D call).
Fix it by implementing native frame copying for Java
VideoRenderer.I420Frame implementation.
BUG=4248
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40639004
Cr-Commit-Position: refs/heads/master@{#8252}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8252 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| cc64a9cc4fcc7df95cee0fc069b8924c3fb196ce |
05-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
As of r8230 (https://webrtc-codereview.appspot.com/39739004/) a new Echo Delay Metric was added calculating the fraction of poor values that may cause the AEC to fail. There are currently two methods for GetDelayMetrics() in webrtc::AutioProcessing and one is deprecated.
This CL updates
- GetEcDelayMetrics()
- voe_auto_test
- talk/media/(fake)webrtcvoiceengine
BUG=N/A
TESTED=locally and trybots
R=pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41749004
Cr-Commit-Position: refs/heads/master@{#8251}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8251 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| 877ac765ad30a22148da41695fa607682af4a191 |
04-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38909004
Cr-Commit-Position: refs/heads/master@{#8245}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| 520a69e8ea71e93528f258b1c2f85d1660fe9647 |
04-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 8238 "Add RefCounting for TransportProxies"
Failing on Mac64_Debug
> Add RefCounting for TransportProxies
>
> BUG=1574
> R=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/37869004
TBR=decurtis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37159004
Cr-Commit-Position: refs/heads/master@{#8243}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8243 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
|
| c5f697135e626044b15eacdc82fd840fbe74b351 |
04-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 8237 "Cleanup and prepare for bundling."
libjingle_peerconnection_objc_test consistently failing on Mac64 Debug.
> Cleanup and prepare for bundling.
>
> - Add a GetOptions function. Needed for eventual bundle testing to
> confirm that channel options are preserved.
> - Simplify unit tests and cleanup unused code.
>
> BUG=1574
> R=pthatcher@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/39699004
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34959004
Cr-Commit-Position: refs/heads/master@{#8241}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| e2506670a4e57cbd351141d8ccf7635ffd2db093 |
04-Feb-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Add RefCounting for TransportProxies
BUG=1574
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37869004
Cr-Commit-Position: refs/heads/master@{#8238}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8238 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
|
| af01d93aa2d75b39cdcaadd682c5c60336c75ea7 |
04-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
BUG=1574
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39699004
Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| 322a564f49d9c995cfffbaabd3d8c5d5aa326e86 |
03-Feb-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Fix datachannel stats id and timestamp.
Makes the id now be "datachannel_#####" where '####' is the id number for the datachannel.
Adds a timestamp to the data channel reports.
Implements unit tests to verify that the timestamp is set correctly.
BUG=1805
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33119004
Cr-Commit-Position: refs/heads/master@{#8236}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8236 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 0e81fdf5d2c2665bc3d23e07cfd9ea7f7d36aed9 |
03-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40569004
Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/rtputils.cc
edia/base/rtputils.h
edia/base/rtputils_unittest.cc
ession/media/channel_unittest.cc
|
| 19f3f71c9873cf5f6d647becd3620ddf8fd6ba7c |
02-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Fix apparent typo: int -> char.
The surrounding similar methods all used unsigned char, using unsigned int in
this case looks like an accident, especially since the function passes on the
value in question to a function expecting a uint8.
BUG=none
TEST=none
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40529004
Cr-Commit-Position: refs/heads/master@{#8228}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| 026b892e724c3f47bde92d773d84099768e57ec8 |
30-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Using << on an int8_t or uint8_t will output a character rather than a number.
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 005b6fffe639b50ba2deebe32424e109fd40f2b1 |
30-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
BUG=none
TEST=none
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39649004
Cr-Commit-Position: refs/heads/master@{#8222}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8222 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface_unittest.cc
|
| 5e161616b17900c06809e7275afca96363d44ad5 |
30-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove CPU monitor from WebRtcVideoEngine2.
CPU adaptation is based on timings done inside webrtc, not actual CPU
values anymore. This code has never been wired up and is causing flakes
on at least valgrind, but possibly also on actual platforms.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34089004
Cr-Commit-Position: refs/heads/master@{#8221}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8221 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| aef0779dab1760951def1bdbb3f49835a8189293 |
30-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Rewrite ThreadWindows.
* Remove "dead" and "alive" variables.
* Remove critical section
* Skip synchronizing with the worker thread to verify startup (no need).
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.
Also added some TODOs for myself for the ThreadWrapper interface.
I'm removing the HasNoMonitorThread test since it is no longer relevant and ends up checking the wrong thing (ProcessThread - a generic thread type) in the wrong way (parsing a debug log) :) I think it served a purpose some years ago, but things have changed since.
BUG=2902
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37069004
Cr-Commit-Position: refs/heads/master@{#8220}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8220 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 8820ac7cc44f8777b6b2372ce4bfd0bde0ae0289 |
30-Jan-2015 |
braveyao@webrtc.org <braveyao@webrtc.org> |
peerconnectin_server: missing comma in sprintfn() in r8128
BUG=4244
TEST=Manual Test
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37079004
Cr-Commit-Position: refs/heads/master@{#8213}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8213 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/server/peer_channel.cc
|
| 50fe359eb614e1bbe41124b9c19263019da0395d |
29-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add tracing for slow paths in new video API.
Allows tracking what actually takes time in SetRemoteDescription and
SetLocalDescription.
BUG=1788
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38809004
Cr-Commit-Position: refs/heads/master@{#8202}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8202 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 4161715e3f7e744bc9ef3d3ae437da1e8e4de38d |
29-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Remove ChangeUniqueID.
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcpassthroughrender.h
|
| a26f511dd2300d6d40052490d9ad7684a5590658 |
29-Jan-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove frame copy in ViEExternalRendererImpl::RenderFrame
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
BUG=1128,4227
R=mflodman@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8136
Review URL: https://webrtc-codereview.appspot.com/36489004
Cr-Commit-Position: refs/heads/master@{#8199}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8199 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| a742cb1f37aefad358dab5cec8d1b80109db12b2 |
29-Jan-2015 |
braveyao@webrtc.org <braveyao@webrtc.org> |
Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off.
BUG=3872
TEST=Manual Test
R=jiayl@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36989004
Cr-Commit-Position: refs/heads/master@{#8193}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8193 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
xamples/peerconnection/server/server_test.html
|
| e7a4a12f83b342f1c2c455366ce465f07a9330b1 |
28-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Add arraysize() macro from Chromium, and make use of it in a few places.
This not only shortens some test code, it makes it more robust against changing
the lengths of the arrays later and forgetting to update the length constants
(which bit me).
BUG=none
TEST=none
R=hta@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34829004
Cr-Commit-Position: refs/heads/master@{#8191}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8191 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| a67ca1a3bb69d5237ce7cf8e62ceb5ad37c49785 |
28-Jan-2015 |
honghaiz@google.com <honghaiz@google.com> |
Only report the first rtp packet because it indicates the media has started flowing.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37829004
Cr-Commit-Position: refs/heads/master@{#8189}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8189 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel.cc
|
| 36401aba6227c9be36cfc15fe6a5d981ecc78a95 |
27-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Update GAE API paths for join/leave.
BUG=4221
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33069004
Cr-Commit-Position: refs/heads/master@{#8174}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8174 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/objc/AppRTCDemo/ARDAppClient+Internal.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/ARDJoinResponse+Internal.h
xamples/objc/AppRTCDemo/ARDJoinResponse.h
xamples/objc/AppRTCDemo/ARDJoinResponse.m
xamples/objc/AppRTCDemo/ARDRegisterResponse+Internal.h
xamples/objc/AppRTCDemo/ARDRegisterResponse.h
xamples/objc/AppRTCDemo/ARDRegisterResponse.m
xamples/objc/AppRTCDemo/ARDRoomServerClient.h
xamples/objc/AppRTCDemo/tests/ARDAppClientTest.mm
ibjingle_examples.gyp
|
| fc5ad95fecc5ddc7d98dcfbac1c4e75a7814253f |
27-Jan-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139
Link to original CL: https://review.webrtc.org/36909004/
R=pbos@webrtc.org
TBR=pthatcher@webrtc.org
BUG=4227
Review URL: https://webrtc-codereview.appspot.com/39669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8162 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideorenderer.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 8501ee632bc4885acdd5e7732d2784b10e68d7ff |
27-Jan-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Support VP8 HW decoding on devices with Exynos codec.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 82415e395f02e5dc0a44a5447eee1aa6b52b766e |
26-Jan-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Update AppRTCDemo to use renamed GAE messages.
BUG=4221
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8158 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 7519de519e8bafb8278b47a8c88444a2209487f9 |
23-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
> Remove frame copy in ViEExternalRendererImpl::RenderFrame
>
> Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
>
> BUG=1128
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36489004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8144 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 0f988447496e5d656d52bea279c8511d3569cb11 |
23-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Revert 8139 "Implement elapsed time and capture start NTP time e..."
> Implement elapsed time and capture start NTP time estimation.
>
> These two elements are required for end-to-end delay estimation.
>
> BUG=1788
> R=stefan@webrtc.org
> TBR=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36909004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8143 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideorenderer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| dacdd9403d30cdb13ab2de645841edd2ae76950d |
23-Jan-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Reland r7980:
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| ad3ee2c46bf502a18847229d42dd081c9e753c70 |
23-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement elapsed time and capture start NTP time estimation.
These two elements are required for end-to-end delay estimation.
BUG=1788
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8139 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideorenderer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| a02d76845f266ec692af0854aca433e8246d7715 |
23-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
Disabling the test on all platforms since it's likely it can happen
on any platform, even if it's only been observed on Win x64 Release.
Running tests in parallel is a huge performance benefit to the team,
since it approximately reduces build cycle with 60-75%.
BUG=4219
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8138 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtmfsender_unittest.cc
|
| 182ea46facde45811faebec40ad4981fd8db56a1 |
23-Jan-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove frame copy in ViEExternalRendererImpl::RenderFrame
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
BUG=1128
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8136 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 586f2eda0d90b84ffefdf2c3662073f22af73bdb |
23-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Change GetStreamBySsrc to not copy StreamParams.
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :)
BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/rtpdataengine.cc
edia/base/streamparams.cc
edia/base/streamparams.h
ession/media/bundlefilter.cc
ession/media/channel.cc
ession/media/mediasession.cc
ession/media/mediasession.h
|
| b40c7bb53c35460d32588c9661bf566681beaf1d |
22-Jan-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Change sprintf use in talk samples to snprintf
BUG=2301
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8128 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/server/peer_channel.cc
xamples/peerconnection/server/utils.cc
|
| cfd82dfc1156f6610388bec0ebbdeacaf47e9719 |
22-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| cceb166a3fd2724c679da7d093149b0511e8d99b |
22-Jan-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix a use-after-free when sending queued messages is aborted for blocked channel.
BUG=4187
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8119 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
|
| 4fb7e2584326050a707aef544028fa9cb616ec89 |
21-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Update StatsReport and by extension StatsCollector to reduce data copying.
Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.
BUG=2822
R=hta@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| fedb9ea6bcd49223933574b1386753c658a1789c |
21-Jan-2015 |
braveyao@webrtc.org <braveyao@webrtc.org> |
Correct the class name in peerconnection_jni.cc.
BUG=4194
TEST=Manual Test
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8106 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 5f93d0a140515e3b8cdd1b9a4c6f5871144e5dee |
20-Jan-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
pp/webrtc/androidtests/src/org/webrtc/PeerConnectionAndroidTest.java
pp/webrtc/audiotrack.cc
pp/webrtc/audiotrack.h
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/dtmfsender.cc
pp/webrtc/dtmfsender.h
pp/webrtc/dtmfsender_unittest.cc
pp/webrtc/dtmfsenderinterface.h
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/AudioSource.java
pp/webrtc/java/src/org/webrtc/AudioTrack.java
pp/webrtc/java/src/org/webrtc/DataChannel.java
pp/webrtc/java/src/org/webrtc/IceCandidate.java
pp/webrtc/java/src/org/webrtc/Logging.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/MediaConstraints.java
pp/webrtc/java/src/org/webrtc/MediaSource.java
pp/webrtc/java/src/org/webrtc/MediaStream.java
pp/webrtc/java/src/org/webrtc/MediaStreamTrack.java
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/java/src/org/webrtc/SdpObserver.java
pp/webrtc/java/src/org/webrtc/SessionDescription.java
pp/webrtc/java/src/org/webrtc/StatsObserver.java
pp/webrtc/java/src/org/webrtc/StatsReport.java
pp/webrtc/java/src/org/webrtc/VideoCapturer.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/src/org/webrtc/VideoSource.java
pp/webrtc/java/src/org/webrtc/VideoTrack.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
pp/webrtc/jsep.h
pp/webrtc/jsepicecandidate.cc
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/jsepsessiondescription.h
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource.h
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastream.cc
pp/webrtc/mediastream.h
pp/webrtc/mediastream_unittest.cc
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamproxy.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/mediastreamtrack.h
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/notifier.h
pp/webrtc/objc/RTCAudioTrack+Internal.h
pp/webrtc/objc/RTCAudioTrack.mm
pp/webrtc/objc/RTCDataChannel+Internal.h
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/objc/RTCEAGLVideoView.m
pp/webrtc/objc/RTCEnumConverter.h
pp/webrtc/objc/RTCEnumConverter.mm
pp/webrtc/objc/RTCI420Frame+Internal.h
pp/webrtc/objc/RTCI420Frame.mm
pp/webrtc/objc/RTCICECandidate+Internal.h
pp/webrtc/objc/RTCICECandidate.mm
pp/webrtc/objc/RTCICEServer+Internal.h
pp/webrtc/objc/RTCICEServer.mm
pp/webrtc/objc/RTCMediaConstraints+Internal.h
pp/webrtc/objc/RTCMediaConstraints.mm
pp/webrtc/objc/RTCMediaConstraintsNative.cc
pp/webrtc/objc/RTCMediaConstraintsNative.h
pp/webrtc/objc/RTCMediaSource+Internal.h
pp/webrtc/objc/RTCMediaSource.mm
pp/webrtc/objc/RTCMediaStream+Internal.h
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCMediaStreamTrack+Internal.h
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCNSGLVideoView.m
pp/webrtc/objc/RTCOpenGLVideoRenderer.mm
pp/webrtc/objc/RTCPair.m
pp/webrtc/objc/RTCPeerConnection+Internal.h
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/RTCSessionDescription+Internal.h
pp/webrtc/objc/RTCSessionDescription.mm
pp/webrtc/objc/RTCStatsReport+Internal.h
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/objc/RTCVideoCapturer+Internal.h
pp/webrtc/objc/RTCVideoCapturer.mm
pp/webrtc/objc/RTCVideoRendererAdapter.h
pp/webrtc/objc/RTCVideoRendererAdapter.mm
pp/webrtc/objc/RTCVideoSource+Internal.h
pp/webrtc/objc/RTCVideoSource.mm
pp/webrtc/objc/RTCVideoTrack+Internal.h
pp/webrtc/objc/RTCVideoTrack.mm
pp/webrtc/objc/public/RTCAudioSource.h
pp/webrtc/objc/public/RTCAudioTrack.h
pp/webrtc/objc/public/RTCDataChannel.h
pp/webrtc/objc/public/RTCEAGLVideoView.h
pp/webrtc/objc/public/RTCI420Frame.h
pp/webrtc/objc/public/RTCICECandidate.h
pp/webrtc/objc/public/RTCICEServer.h
pp/webrtc/objc/public/RTCMediaConstraints.h
pp/webrtc/objc/public/RTCMediaSource.h
pp/webrtc/objc/public/RTCMediaStream.h
pp/webrtc/objc/public/RTCMediaStreamTrack.h
pp/webrtc/objc/public/RTCNSGLVideoView.h
pp/webrtc/objc/public/RTCOpenGLVideoRenderer.h
pp/webrtc/objc/public/RTCPair.h
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/objc/public/RTCPeerConnectionDelegate.h
pp/webrtc/objc/public/RTCPeerConnectionFactory.h
pp/webrtc/objc/public/RTCSessionDescription.h
pp/webrtc/objc/public/RTCSessionDescriptionDelegate.h
pp/webrtc/objc/public/RTCStatsDelegate.h
pp/webrtc/objc/public/RTCStatsReport.h
pp/webrtc/objc/public/RTCTypes.h
pp/webrtc/objc/public/RTCVideoCapturer.h
pp/webrtc/objc/public/RTCVideoRenderer.h
pp/webrtc/objc/public/RTCVideoSource.h
pp/webrtc/objc/public/RTCVideoTrack.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.h
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.m
pp/webrtc/objctests/mac/main.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/portallocatorfactory.cc
pp/webrtc/portallocatorfactory.h
pp/webrtc/proxy.h
pp/webrtc/proxy_unittest.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/remotevideocapturer.cc
pp/webrtc/remotevideocapturer.h
pp/webrtc/remotevideocapturer_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/streamcollection.h
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/test/fakedtlsidentityservice.h
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
pp/webrtc/test/testsdpstrings.h
pp/webrtc/umametrics.h
pp/webrtc/videosource.cc
pp/webrtc/videosource.h
pp/webrtc/videosource_unittest.cc
pp/webrtc/videosourceinterface.h
pp/webrtc/videosourceproxy.h
pp/webrtc/videotrack.cc
pp/webrtc/videotrack.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp.h
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
uild/build_jar.sh
uild/common.gypi
uild/isolate.gypi
uild/objc_app.gypi
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/AppRTCProximitySensor.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/SettingsFragment.java
xamples/android/src/org/appspot/apprtc/UnhandledExceptionHandler.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/src/org/appspot/apprtc/util/AppRTCUtils.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/android/src/org/appspot/apprtc/util/LooperExecutor.java
xamples/androidtests/src/org/appspot/apprtc/test/LooperExecutorTest.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
xamples/objc/AppRTCDemo/ARDAppClient+Internal.h
xamples/objc/AppRTCDemo/ARDAppClient.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.h
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/ARDCEODTURNClient.h
xamples/objc/AppRTCDemo/ARDCEODTURNClient.m
xamples/objc/AppRTCDemo/ARDMessageResponse+Internal.h
xamples/objc/AppRTCDemo/ARDMessageResponse.h
xamples/objc/AppRTCDemo/ARDMessageResponse.m
xamples/objc/AppRTCDemo/ARDRegisterResponse+Internal.h
xamples/objc/AppRTCDemo/ARDRegisterResponse.h
xamples/objc/AppRTCDemo/ARDRegisterResponse.m
xamples/objc/AppRTCDemo/ARDRoomServerClient.h
xamples/objc/AppRTCDemo/ARDSignalingChannel.h
xamples/objc/AppRTCDemo/ARDSignalingMessage.h
xamples/objc/AppRTCDemo/ARDSignalingMessage.m
xamples/objc/AppRTCDemo/ARDTURNClient.h
xamples/objc/AppRTCDemo/ARDUtilities.h
xamples/objc/AppRTCDemo/ARDUtilities.m
xamples/objc/AppRTCDemo/ARDWebSocketChannel.h
xamples/objc/AppRTCDemo/ARDWebSocketChannel.m
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.h
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.m
xamples/objc/AppRTCDemo/RTCICEServer+JSON.h
xamples/objc/AppRTCDemo/RTCICEServer+JSON.m
xamples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h
xamples/objc/AppRTCDemo/RTCMediaConstraints+JSON.m
xamples/objc/AppRTCDemo/RTCSessionDescription+JSON.h
xamples/objc/AppRTCDemo/RTCSessionDescription+JSON.m
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.h
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.m
xamples/objc/AppRTCDemo/ios/ARDMainView.h
xamples/objc/AppRTCDemo/ios/ARDMainView.m
xamples/objc/AppRTCDemo/ios/ARDMainViewController.h
xamples/objc/AppRTCDemo/ios/ARDMainViewController.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
xamples/objc/AppRTCDemo/ios/AppRTCDemo-Prefix.pch
xamples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.h
xamples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.m
xamples/objc/AppRTCDemo/ios/main.m
xamples/objc/AppRTCDemo/mac/APPRTCAppDelegate.h
xamples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m
xamples/objc/AppRTCDemo/mac/APPRTCViewController.h
xamples/objc/AppRTCDemo/mac/APPRTCViewController.m
xamples/objc/AppRTCDemo/mac/main.m
xamples/objc/AppRTCDemo/tests/ARDAppClientTest.mm
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
xamples/peerconnection/client/defaults.cc
xamples/peerconnection/client/defaults.h
xamples/peerconnection/client/flagdefs.h
xamples/peerconnection/client/linux/main.cc
xamples/peerconnection/client/linux/main_wnd.cc
xamples/peerconnection/client/linux/main_wnd.h
xamples/peerconnection/client/main.cc
xamples/peerconnection/client/main_wnd.cc
xamples/peerconnection/client/main_wnd.h
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/client/peer_connection_client.h
xamples/peerconnection/server/data_socket.cc
xamples/peerconnection/server/data_socket.h
xamples/peerconnection/server/main.cc
xamples/peerconnection/server/peer_channel.cc
xamples/peerconnection/server/peer_channel.h
xamples/peerconnection/server/utils.cc
xamples/peerconnection/server/utils.h
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_tests.gyp
ibjingle_unittest.isolate
edia/base/capturemanager_unittest.cc
edia/base/capturerenderadapter.cc
edia/base/capturerenderadapter.h
edia/base/device.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/hybriddataengine.h
edia/base/mediaengine.cc
edia/base/screencastid.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturerfactory.h
edia/base/videocommon.cc
edia/base/videocommon.h
edia/base/videoengine_unittest.h
edia/base/videoframefactory.cc
edia/base/videoframefactory.h
edia/base/yuvframegenerator.cc
edia/base/yuvframegenerator.h
edia/devices/carbonvideorenderer.cc
edia/devices/carbonvideorenderer.h
edia/devices/filevideocapturer.cc
edia/devices/filevideocapturer.h
edia/devices/gdivideorenderer.cc
edia/devices/gdivideorenderer.h
edia/devices/gtkvideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/devices/macdevicemanagermm.mm
edia/devices/v4llookup.cc
edia/devices/v4llookup.h
edia/devices/videorendererfactory.h
edia/devices/yuvframescapturer.cc
edia/devices/yuvframescapturer.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/constants.h
edia/webrtc/fakewebrtcdeviceinfo.h
edia/webrtc/fakewebrtcvcmfactory.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/simulcast.h
edia/webrtc/webrtccommon.h
edia/webrtc/webrtcexport.h
edia/webrtc/webrtcpassthroughrender_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideocapturer_unittest.cc
edia/webrtc/webrtcvideocapturerfactory.h
edia/webrtc/webrtcvideodecoderfactory.h
edia/webrtc/webrtcvideoencoderfactory.h
edia/webrtc/webrtcvideoframefactory.h
edia/webrtc/webrtcvie.h
edia/webrtc/webrtcvoe.h
ession/media/channel_unittest.cc
ession/media/channelmanager_unittest.cc
ession/media/mediarecorder_unittest.cc
ession/media/planarfunctions_unittest.cc
ession/media/rtcpmuxfilter_unittest.cc
|
| 853049fa308c3181769e8ee13eddb289d573a065 |
20-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Move internal capture+render to build_with_chromium==0 condition
This will avoid errors related to DirectX not being found
for Chromium bots (mainly GN, but it's safest to do the same
changes for GYP since they also make sense there as GYP generation
go slightly faster without having to process those targets).
Thanks to vchigrin@yandex-team.ru for originally suggesting
this being fixed in
https://webrtc-codereview.appspot.com/37639004/
TESTED=
Successfully ran:
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dbuild_with_chromium=1
and trybots.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8102 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 8e327c45d0940fd5bc46c3fe8d24363be07706ac |
19-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8095
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 43e54e36bff3f6159e9c7ac0aa40beafca485c56 |
19-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8095 "Update StatsCollector's interface in preparation of..."
> Update StatsCollector's interface in preparation of more changes.
>
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
>
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
>
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
>
> BUG=2822
> R=perkj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36829004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 5b76fd79dfbfa78f1b034c0698771298cd15f175 |
19-Jan-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| f9d3555ec384a4ed428a114f5fd33abaefce30c6 |
19-Jan-2015 |
phoglund@webrtc.org <phoglund@webrtc.org> |
Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.
The was was really, really difficult to run before because you needed
a custom env with both LD_PRELOAD and library path. Now the script will
set up the correct library path and be more transparent about what it
requires.
BUG=None
TESTED=locally
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8093 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
|
| ff9462eb540023b1ff8d0fda860504121a3b6f8a |
19-Jan-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.
Tests are flaky on tsan, disabling for now.
BUG=4135
R=kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8089 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 487a4442152e2c70146aa2d2c6ccb370233c056c |
15-Jan-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Add stats collection for the data channel.
BUG=1805
R=bemasc@chromium.org, hta@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannelinterface.h
pp/webrtc/mediastreamsignaling.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtcsession.h
|
| ef2a5dd3983152365102881af01942629dc720d6 |
15-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Update AppRTCDemo UI.
- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCVideoTrack.mm
xamples/objc/AppRTCDemo/ios/APPRTCAppDelegate.h
xamples/objc/AppRTCDemo/ios/APPRTCAppDelegate.m
xamples/objc/AppRTCDemo/ios/APPRTCViewController.h
xamples/objc/AppRTCDemo/ios/APPRTCViewController.m
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.h
xamples/objc/AppRTCDemo/ios/ARDAppDelegate.m
xamples/objc/AppRTCDemo/ios/ARDMainView.h
xamples/objc/AppRTCDemo/ios/ARDMainView.m
xamples/objc/AppRTCDemo/ios/ARDMainViewController.h
xamples/objc/AppRTCDemo/ios/ARDMainViewController.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallView.m
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.h
xamples/objc/AppRTCDemo/ios/ARDVideoCallViewController.m
xamples/objc/AppRTCDemo/ios/Default.png
xamples/objc/AppRTCDemo/ios/Info.plist
xamples/objc/AppRTCDemo/ios/ResourceRules.plist
xamples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.h
xamples/objc/AppRTCDemo/ios/UIImage+ARDUtilities.m
xamples/objc/AppRTCDemo/ios/en.lproj/APPRTCViewController.xib
xamples/objc/AppRTCDemo/ios/main.m
xamples/objc/AppRTCDemo/ios/resources/Default-568h.png
xamples/objc/AppRTCDemo/ios/resources/Roboto-Regular.ttf
xamples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp.png
xamples/objc/AppRTCDemo/ios/resources/ic_call_end_black_24dp@2x.png
xamples/objc/AppRTCDemo/ios/resources/ic_clear_black_24dp.png
xamples/objc/AppRTCDemo/ios/resources/ic_clear_black_24dp@2x.png
ibjingle_examples.gyp
|
| 61c1247224e2b696b10303b0b5479b3a246f4ff0 |
15-Jan-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Fix a case where empty candidate id is used
BUG=4161
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| fd630a50d23f0c2be2517a354be5456374d20689 |
15-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior.
R=decurtis@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface.h
|
| f1c8b905204bc7a6c74271ead038f5d80d8d3eed |
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEncoderFactory2.
This interface is no longer required and just adds complexity.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/33009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| f18fba2f7b3d1fad7b7b38a9a5dc281bef06c50e |
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement SimulcastEncoderAdapter support.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/37589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 8315d7de8551963c53162e320835c158088fcdd6 |
14-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove dual stream functionality in VoiceEngine
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.
BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| b4e5d1b34e1e8edd5b324cd8419726cef66cd0af |
14-Jan-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove RTX SSRC when deleting the default receive stream.
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 2ebfac5649a5e48fbbc501b42a4336ff979c03e6 |
14-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Remove COMPILE_ASSERT and use static_assert everywhere
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 86e1e487e73ec33177d8c03989042a31cc157575 |
14-Jan-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Move system_wrappers.gyp files to the proper directory.
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| ef090927f48bc9144be6b724df69c2c09119766e |
14-Jan-2015 |
phoglund@webrtc.org <phoglund@webrtc.org> |
No longer asserting in mocks, split first test case in two methods.
This way assertions will be caught in the test runner instead of crashing other Android threads.
BUG=None
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8054 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
|
| 3df38b442f6ba29722049b4c4d7121053003a1f8 |
13-Jan-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Unify the two copies of compile_assert.h
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| be40eb05795049271d708140d2c79da6246abf6f |
12-Jan-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Allow 720x1280 frames encoding on Android.
Current maximum encoder width and height for Android is
hard-coded to 1280x720, so if device is rotated to portrait
orientation only part 720x1280 camera frame is extracted and
scaled to 1280x720. Increasing maximum height to 1280 allows
feeding video encoder with rotated 720x1280 frames directly
without scaling.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8042 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
|
| 81134d019d1e3a9f7bfb4dc51d90e66e8d06b27d |
12-Jan-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory.
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory.
This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.
Note that both Chrome and the Android implementation use an external signaling thread.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactoryproxy.h
pp/webrtc/proxy.h
ibjingle.gyp
|
| 8f27fcce79584378da97f0d84574564799e138d6 |
09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 8028 "Support associated payload type when registering Rt..."
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com>.
TBR=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 80452d70cb9efed9d891c1f36674559322a075ca |
09-Jan-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Sync Android AppRTCDemo with internal repo.
- Fixed some Lint warnings.
- Switch to OPUS by default.
- Add check to WebSocket connection that public methods are called
on correct thread.
R=jiayl@webrtc.org, wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8032 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/src/org/appspot/apprtc/util/AppRTCUtils.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/android/src/org/appspot/apprtc/util/LooperExecutor.java
|
| 9657265f391cfe473a61b18a4579bbbeb44c9bd8 |
09-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
This reverts r7980.
It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.
Review URL: https://webrtc-codereview.appspot.com/41429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
|
| 2a169640a3225a559f926fe74f1fe1af239e191f |
09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Support associated payload type when registering Rtx payload type.
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.
BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26259004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 2ead571fb6e06b90053b0ee920505fd589b229aa |
08-Jan-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Hard define the GUID for AudioEndpoint to avoid conflicts during compile.
BUG=3996
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8026 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/win32devicemanager.cc
|
| 59062d5aefad3499e89049b60c7a484944253c1b |
07-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Rename SendAndReceiveH264SvcQqvga to VP8 instead.
This looks like it's been incorrect for a while, this test configures
VP8 in QQVGA.
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8018 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 8af11042cb1157b722b55304c292f3091290da4d |
07-Jan-2015 |
decurtis@webrtc.org <decurtis@webrtc.org> |
Avoid reading past end of string in GetLine.
BUG=3881
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8017 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| bab79951ca057d41e2e9a28a03c057b99ed46092 |
07-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Convert FileMediaEngineTest to use more expects.
Allows pinpointing more precisely where a failure occurs.
BUG=4144
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8015 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/filemediaengine_unittest.cc
|
| 07c83a13857c11e324a8966914c2ca30be365114 |
07-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2)
In https://webrtc-codereview.appspot.com/35669004/ the wrong
define was used (OS_WIN only exists in Chromium code).
BUG=4135
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8008 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 4e5115ae73098cdb60c2d442777c169d2a6c2925 |
07-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
There should be no change in behavior, since this is what the default
constructor does.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8007 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnectionFactory.mm
|
| f6a97147602776c672d757e2d598f8643ce6d339 |
06-Jan-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Remove peer connection and signaling calls from UI thread.
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.
R=jiayl@webrtc.org, wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/assets/channel.html
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/AppRTCProximitySensor.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java
xamples/android/src/org/appspot/apprtc/util/LooperExecutor.java
xamples/androidtests/src/org/appspot/apprtc/test/LooperExecutorTest.java
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
ibjingle_examples.gyp
|
| d95435c17ae13670b3a41ee6153a93c5f6eb9118 |
06-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win
These tests have turned out to be flaky on Windows.
BUG=4135
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8004 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| cbe7ca8796a3e0c6b56910640e76943ba4224118 |
06-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision 8e72e1d..271c6cc (307131:309333)
This enables OpenSSL by default for Windows, see
https://chromium.googlesource.com/chromium/src/+/8e72e1d..271c6cc/build/common.gypi
which required libjingle_tests.gyp to be updated since the
targets in third_party/nss/nss.gyp was moved into a condition in
https://codereview.chromium.org/694643002.
New Android dependencies are required due to being introduced in
build/android/pylib/remote/device/remote_device_test_run.py
of https://chromium.googlesource.com/chromium/src/+/5c49978f095340a59c62faaafe02a9527ec7728b
This should also fix Android test execution that started failing after
https://codereview.chromium.org/815213002 was submitted, since
it's based on https://chromium.googlesource.com/chromium/src/+/e2a338fac902ff391f761c67580b8de00d4adfdf
Relevant other changes:
* src/buildtools: 535aff2..23a4e2f
* src/third_party/android_tools: 4f723e2..8fe116f
* src/third_party/boringssl/src: 00505ec..306e520
* src/third_party/icu: 53ecf0f..51c1a4c
* src/third_party/libvpx: 9fbec81..d3f3dce
* src/tools/swarming_client: 1d4965c..119b084
Details: https://chromium.googlesource.com/chromium/src/+/8e72e1d..271c6cc/DEPS
Clang version updated 218707:223108:
https://chromium.googlesource.com/chromium/src/+/8e72e1d..271c6cc/tools/clang/scripts/update.sh
Due to this, we had to disable deadlock detection for TSan
due to a bug in Clang (see webrtc:
BUG=4106
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8003 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 3a63a3c35d93616606c23d27583b83a198b94a3e |
06-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
iOS AppRTC: First unit test.
Tests basic session ICE connection by stubbing out network components, which have been refactored to faciliate testing.
BUG=3994
R=jiayl@webrtc.org, kjellander@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8002 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/ios_test.plist
uild/ios_tests.gypi
uild/objc_app.gypi
uild/objc_app.plist
xamples/objc/AppRTCDemo/ARDAppClient+Internal.h
xamples/objc/AppRTCDemo/ARDAppClient.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDAppEngineClient.h
xamples/objc/AppRTCDemo/ARDAppEngineClient.m
xamples/objc/AppRTCDemo/ARDCEODTURNClient.h
xamples/objc/AppRTCDemo/ARDCEODTURNClient.m
xamples/objc/AppRTCDemo/ARDMessageResponse+Internal.h
xamples/objc/AppRTCDemo/ARDMessageResponse.m
xamples/objc/AppRTCDemo/ARDRegisterResponse+Internal.h
xamples/objc/AppRTCDemo/ARDRegisterResponse.m
xamples/objc/AppRTCDemo/ARDRoomServerClient.h
xamples/objc/AppRTCDemo/ARDSignalingChannel.h
xamples/objc/AppRTCDemo/ARDTURNClient.h
xamples/objc/AppRTCDemo/ARDWebSocketChannel.h
xamples/objc/AppRTCDemo/ARDWebSocketChannel.m
xamples/objc/AppRTCDemo/ios/APPRTCViewController.m
xamples/objc/AppRTCDemo/mac/APPRTCViewController.m
xamples/objc/AppRTCDemo/tests/ARDAppClientTest.mm
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| c37e72e890cb1c769af9006dbd2e582c1a2e2a50 |
05-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Make setting identical RTP extensions a no-op.
Setting extensions are responsible for a lot of stream tear-downs
causing substantial slowdowns in SetRemoteDescription.
BUG=1788,4077
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7998 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 433006a6c230cdc051446e252af658b671d0bd20 |
05-Jan-2015 |
wzh@webrtc.org <wzh@webrtc.org> |
Fixed style issues from lint and got rid of unused fields.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7995 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCProximitySensor.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
|
| 8390c2762e6be87e213c1cb0d65af480c52e2e39 |
02-Jan-2015 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add two unit tests for Android AppRTCDemo.
First unit test will create peer connection client, run
for a few second, close it and verify that there were
no any errors and local video was rendered.
Second unit test will run peer connection in a loopback mode.
To run the test from command line install AppRTCDemoTest.apk
and execute the command:
adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7991 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/README
xamples/androidtests/AndroidManifest.xml
xamples/androidtests/README
xamples/androidtests/ant.properties
xamples/androidtests/build.xml
xamples/androidtests/project.properties
xamples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
ibjingle_examples.gyp
|
| 896888b7e4cea97c65786b0e63bf2f65dc7d2390 |
02-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove min bitrate from simulcast streams.
Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/38529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/simulcast.cc
edia/webrtc/simulcast.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 9eacb8cc5911eb38d7f31d0cfe07bde981d33316 |
02-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Make P2PTestConductor use VirtualSocketServer.
Permits running JsepPeerConnectionP2PTestClient in parallel.
TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*
Review URL: https://webrtc-codereview.appspot.com/37459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| c62749fb471a9bf30b05a270c532c49ebea2f03d |
02-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Parallelize MediaRecorder unittests.
Exchanging static filenames for temporary ones, permitting tests to be
run in parallel without conflicting parallel uses of the same filenames.
TBR=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 -r 100 out/Debug/libjingle_p2p_unittest
Review URL: https://webrtc-codereview.appspot.com/34589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7987 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/mediarecorder_unittest.cc
|
| 27f53175605aa4546d1e19a83529f445718f94ea |
31-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Use the prod GAE server in AppRTCDemo for iOS.
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7985 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/objc/AppRTCDemo/ARDAppClient.m
|
| 5eb71eb4f470bac0cbae0e1be4db8c83bc16fcd9 |
30-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix style issues from lint.
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7984 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/SettingsFragment.java
|
| b2bda67497d9da69221c5814d2b815086c04e0ca |
30-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Removing old channel code from a few more places.
Plus adding peer connection close event.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7982 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
|
| c5fd66dcdfdba3ec114cc5b5c0337eba503cee40 |
29-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Accept incoming pings before remote answer is set to reduce connection latency.
BUG=4068
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
|
| b024da31225ef47470f09798dd745ab99840bb95 |
29-Dec-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
Add support for audio device selection in AppRTCDemo.
Summary:
- Creates a list of available (possible to select) audio devices.
- Automatically selects (routes audio) the "best/default" audio device.
- If possible, starts a proximity sensor that will switch between headset earpiece and speaker phone based on how close the a person's ear the mobile device is held.
TBR=glaznev
BUG=4103,4109
Review URL: https://webrtc-codereview.appspot.com/31239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7978 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/AppRTCProximitySensor.java
xamples/android/src/org/appspot/apprtc/util/AppRTCUtils.java
ibjingle_examples.gyp
|
| 5ad4178137ac869f1e057e07c3a171e11763d9df |
23-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move the Jingle-specific network code into webrtc/libjingle.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/mediasession.cc
|
| 46d4d29a751c559b6f01b311a1e4aa14a2586a46 |
23-Dec-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Add field trial for screenshare bitrates when using temporal layers.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/webrtc/simulcast.cc
edia/webrtc/simulcast.h
edia/webrtc/simulcast_unittest.cc
edia/webrtc/webrtcvideoengine2.cc
|
| 086c8d5a029d95f72a16eabd8a31e24a4213d5dc |
22-Dec-2014 |
braveyao@webrtc.org <braveyao@webrtc.org> |
Use a temporary buffer to scale a screencast in OnFrameCaptured
BUG=3903
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/23909005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7973 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer.cc
|
| 4c0544ab07987fa080a832123bee5e61750fd815 |
19-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository.
Also, fix the includes and header guards of examples/call.
R=juberti@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7972 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
ession/media/call.cc
ession/media/call.h
ession/media/currentspeakermonitor_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
|
| 7ce4a584aa5f6f2d4b62a152d66e7fe821034bf9 |
19-Dec-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Add initWithCoder to RTCEAGLVideoView.
Allows for proper OpenGL initialization if view is created from
storyboard.
BUG=3896
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7970 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCEAGLVideoView.m
|
| a6f7ba6848302d142ba769615d12bbf77a13e6e6 |
19-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Add a AppRTCDemo setting to change the GAE server.
BUG=4041
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7966 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 742386a13670337db6e3bbf4cf54e7cb24a9b717 |
19-Dec-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Enable payload-based padding by default and remove the API.
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 5647877b2d7b299837ed7ab8e8270d593fe5aa79 |
19-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_examples.gyp
ession/media/call.h
ession/media/mediasessionclient.h
|
| aacc23465b72151fece2e6836a7c43463d3ed41d |
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
(This is the 3rd try)
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.h
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.h
|
| 16a05dddb86f2bd29d6c1641224438c1bee13e78 |
18-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7955 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/README
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/GAEChannelClient.java
xamples/android/src/org/appspot/apprtc/GAERTCClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
ibjingle_examples.gyp
|
| f5847d7746df8b640b65a8e47849030adb7a3af2 |
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well.
R=juberti@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7953 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/securetunnelsessionclient.h
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
|
| ce4e9a356200170abcdd44ff2af95f87a6781b8e |
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor some receive-side stats.
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| a9cf079248e1274b2ddf36ab1dc179a2b6eb9deb |
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rename external_hmac_ctx_t to ExternalHmacContext.
_t types are reserved by POSIX.
R=juberti@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/33699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7944 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/externalhmac.cc
ession/media/externalhmac.h
ession/media/srtpfilter.cc
|
| 4cb3856a4d4782cc7abf228a7f01ea70812d9fb1 |
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.
BUG=
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.h
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.h
ession/tunnel/pseudotcpchannel.h
|
| 536f999e58ee7456d116afad734aa64d548f1a49 |
18-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
This is an un-revert of r7992 and r7993.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.h
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.h
ession/tunnel/pseudotcpchannel.h
|
| bc03192560a591ad33c84c707f43710d98e330a3 |
17-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository.
R=juberti@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7936 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/call/Info.plist
xamples/call/call_main.cc
xamples/call/call_unittest.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/console.cc
xamples/call/console.h
xamples/call/friendinvitesendtask.cc
xamples/call/friendinvitesendtask.h
xamples/call/mediaenginefactory.cc
xamples/call/mediaenginefactory.h
xamples/call/muc.h
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.cc
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
ibjingle_examples.gyp
|
| 209df9bf77b68fd872c4218704f98418f7b28ae6 |
17-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Change MockStatsObserver to grab values inside of OnComplete.
This is done since StatsReportCopyable is going away and the list of
supported properties of the mock class is known.
StatsReports holds a list of pointers to objects that cannot be cached,
so this is a simple way to grab the values when they're available.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7932 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/mockpeerconnectionobservers.h
|
| e728ee03ba093ddb9fa6fb803994969801a4f601 |
17-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove or rename typedefs with _t prefixes.
_t prefixes are reserved for additional typenames in POSIX.
R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/36559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/srtpfilter.h
|
| 950c51825109c2ca352317edef0a33777d0e6678 |
17-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Committed: https://code.google.com/p/webrtc/source/detail?r=7906
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| f050791ba071eb208da4e95abc2ff21f57d0738f |
16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
This reverts r7992.
It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.h
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.h
ession/tunnel/pseudotcpchannel.h
|
| 4afb59903c2dcc893cd86a973cc16da4201e387c |
16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.h
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.h
ession/tunnel/pseudotcpchannel.h
|
| e2b7585bc277e211b7d9fc1e3e8046ea41484b5d |
16-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
R=juberti@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
edia/base/streamparams.cc
edia/base/streamparams.h
ession/media/channel.cc
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
|
| 55360ae402908b24757c7983c587e69ea485e9e6 |
16-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Revert "Add adapter_type into Candidate object."
This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.
BUG=
TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| aaf02cc2d4f696345ce0e6d5715f2cfa22aea689 |
16-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcpassthroughrender.h
|
| e2e199b89430fd115e1a8f223cde3b398f3eff52 |
15-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Clean up StatsObserver's OnComplete methods (address TODOs).
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/peerconnectioninterface.h
|
| 032b802a8c7f7c153a38f66175371d3e0df5c52f |
15-Dec-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 82121498-> 82126219
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| dd0601fbcf8d1851bc9df1c6a673fe1d4a7496c9 |
15-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Remove unneeded ctor and add a more practical one
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 69bc5a300fe27448bcb61670f2800d3919ed2975 |
15-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Add thread asserts to StatsCollector.
Also adding a "ForTest" postfix to a test-only method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
|
| fb108b5a28a538862a4157e17de795426d86af1e |
15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r7885.
Breaks compile step of other code where network name of
cricket::Candidate is used.
TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/31229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 18a3896bd28b63fa35168cd6c8d41c8cebaab3dd |
15-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r7886:7887.
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
xamples/call/Info.plist
xamples/call/call_main.cc
xamples/call/call_unittest.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/console.cc
xamples/call/console.h
xamples/call/friendinvitesendtask.cc
xamples/call/friendinvitesendtask.h
xamples/call/mediaenginefactory.cc
xamples/call/mediaenginefactory.h
xamples/call/muc.h
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.cc
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/securetunnelsessionclient.h
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
|
| e575e9c40f7e2aeb28486f6e4b96910bc744c7ec |
14-Dec-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| dee76f3b89b9339699e0321a3afc643ee06afa09 |
12-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Move the obvious/easy Jingle-specific code into webrtc/libjingle.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
xamples/call/Info.plist
xamples/call/call_main.cc
xamples/call/call_unittest.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/console.cc
xamples/call/console.h
xamples/call/friendinvitesendtask.cc
xamples/call/friendinvitesendtask.h
xamples/call/mediaenginefactory.cc
xamples/call/mediaenginefactory.h
xamples/call/muc.h
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.cc
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel_unittest.cc
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/securetunnelsessionclient.h
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
|
| 8c9d79a29d9127d4ff8aa4ae386630c72cfb1808 |
12-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| c57310b982cdce138723de91d7b722f8199834ab |
12-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Switch kStatsValueName* constants to be enums instead of char*.
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| 40b276ea7bc5dfd815c79b93b83bdc9ef24f2cc1 |
12-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup little things found when refactoring.
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/33519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/base/videocommon.h
edia/webrtc/webrtcvideoengine.cc
ession/media/call.cc
ession/media/call.h
|
| 2b19f0631233488e891d9db0d170b637dc8fc464 |
11-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up RTT statistics to webrtc::Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/32249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 13518951e37775dd73de0c5661726db08e83cb8c |
11-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove old_factory from WebRtcVideoEngine.
Minor pending cleanup.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/28239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 128fabaf7b94966aa271fa3d3f13da64a74f5b55 |
11-Dec-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""
Original cl description:
Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| a85307737cc9ea3e79b86daf96d455fca4ad1bb4 |
10-Dec-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 81702493-> 81755413
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/webrtc/simulcast.cc
edia/webrtc/simulcast.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| aa2c342c10bb415be56ed077ae1eae0f31847ec9 |
09-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Add back a constructor to fix FYI build.
TBR=perkj
Review URL: https://webrtc-codereview.appspot.com/24349005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.h
|
| 87776a893546e1a503b679172aa68055fd634f7b |
09-Dec-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
iAppRTCDemo: WebSocket based signaling.
Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.
BUG=
R=glaznev@webrtc.org, jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/objc/AppRTCDemo/APPRTCAppClient.h
xamples/objc/AppRTCDemo/APPRTCAppClient.m
xamples/objc/AppRTCDemo/APPRTCConnectionManager.h
xamples/objc/AppRTCDemo/APPRTCConnectionManager.m
xamples/objc/AppRTCDemo/ARDAppClient.h
xamples/objc/AppRTCDemo/ARDAppClient.m
xamples/objc/AppRTCDemo/ARDMessageResponse.h
xamples/objc/AppRTCDemo/ARDMessageResponse.m
xamples/objc/AppRTCDemo/ARDRegisterResponse.h
xamples/objc/AppRTCDemo/ARDRegisterResponse.m
xamples/objc/AppRTCDemo/ARDSignalingMessage.h
xamples/objc/AppRTCDemo/ARDSignalingMessage.m
xamples/objc/AppRTCDemo/ARDSignalingParams.h
xamples/objc/AppRTCDemo/ARDSignalingParams.m
xamples/objc/AppRTCDemo/ARDUtilities.h
xamples/objc/AppRTCDemo/ARDUtilities.m
xamples/objc/AppRTCDemo/ARDWebSocketChannel.h
xamples/objc/AppRTCDemo/ARDWebSocketChannel.m
xamples/objc/AppRTCDemo/GAEChannelClient.h
xamples/objc/AppRTCDemo/GAEChannelClient.m
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.m
xamples/objc/AppRTCDemo/RTCICEServer+JSON.h
xamples/objc/AppRTCDemo/RTCICEServer+JSON.m
xamples/objc/AppRTCDemo/channel.html
xamples/objc/AppRTCDemo/ios/APPRTCViewController.m
xamples/objc/AppRTCDemo/mac/APPRTCViewController.m
xamples/objc/AppRTCDemo/third_party/SocketRocket/LICENSE
xamples/objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.h
xamples/objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.m
ibjingle_examples.gyp
|
| 0babb4a4e68f60a2862c98bafe4f9a748d077fff |
09-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Fix a comment.
R=juberti@webrtc.org, pbos@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| c9d155faebd5556c9ea86306dc15aa9dac0e13f7 |
09-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Move implementation of types in statstypes. to its cc file.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
|
| a954c07ee1c93175e6ebbeb20517b347474362ae |
09-Dec-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.h
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
|
| 5c3ee4bce6c61bb4095eb3746ba39d3eeab2ee93 |
09-Dec-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Add empty implementation file that will hold statstypes.h implementation.
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.cc
pp/webrtc/statstypes.h
ibjingle.gyp
|
| eef85387ec4a69f10ad102988c9d222f8c69b5da |
09-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix AppRTCDemo closing error for KK and JB Android devices.
- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.
R=jiayl@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
|
| 3b3c4069082e00d0430e75ac242c6f0578e7a528 |
08-Dec-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 7826 "Change Android PeerConnectionUnittest to build usin..."
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575
> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
>
> This also add a new build target to build java PeerConnection using Chromes build macros.
>
> BUG=4031
> R=kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28189004
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| ed7824b1c003a6f5b46ba0c7ce0c9457708c34f2 |
08-Dec-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| e2a9261f3e1d62b628147dc372c6d873f7007dde |
05-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Improve AppRTCDemo connection speed by sending all
http POST requests asynchronously.
R=jiayl@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
|
| bd8cc0b914926e899d607dc07d6e202744ce2795 |
05-Dec-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Add codereview.settings to the /talk subdirectory
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/talk
(which is what you get when working with the repo currently
put in Chrome's src/third_party/libjingle/source/talk).
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7819 4adac7df-926f-26a2-2b94-8c16560cd09d
odereview.settings
|
| 599e299b9dc3dc07fc78cfeaba629566a201b4f1 |
05-Dec-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
cricket::VideoFrame int64 to int64_t.
Needed for successful compile of ios arm64.
BUG=3898
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30359004
Patch from Zeke Chin <tkchin@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 9b5467e88d6f1b3cadb2f9d04941a2b4dec77188 |
05-Dec-2014 |
bemasc@webrtc.org <bemasc@webrtc.org> |
Fix assertion failure when closing data channel, and add a unit test.
BUG=4066
R=jiayl@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7816 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
|
| 4b407aa985e5804a250975d7448aa9fd72b69c29 |
04-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Update AppRTCDemo README with information on 3-dot-apprtc server
and new command line arguments.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7815 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/README
|
| 7169afd9d53fce803858bc954e6cc5ebbf9b1695 |
04-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.
BUG=411086
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/umametrics.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| 369746bcb8daa10dc2686b6317b51e4442a9b9fe |
04-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Support new WebSocket signaling format.
- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.
BUG=3937,3995,4041
R=jiayl@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/GAERTCClient.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
|
| 0fb6ad2004d3b86cb912c93a773e3f9162392e54 |
03-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Check if cpu_monitor_ exists before Stop().
R=asapersson@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| d8aed6b321df136a98f2c3c69cf5391103831b88 |
03-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Verify that cpu_monitor exists before calling Stop().
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7795 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| eb0954248d77bf9a97af27f5905a119c9b8e147a |
03-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Don't reset sequence number for a stream on deactivate/reactivate.
BUG=chromium:431908
R=pbos@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| d01955179a7014f72a381f605ae4d8b0e542c1de |
03-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Change minimum video encoder initialization resolution to
176x144 to ensure HW encoder can be initialized.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| beee9cec22a371eb08a628f73289f36809f3cb54 |
02-Dec-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video.
The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.
TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 146e0fd30f5ff4fb47e0ec8dc824e4d9178c828d |
01-Dec-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Fix the build by putting in a typecast to avoid a comparison between
signed and unsigned ints introduced in cl/81073932.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7776 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| dea5173edfcc6fed0572ff61bbc116918988bd16 |
01-Dec-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add start bitrate and vp8 hw acceleration option to
Android AppRTCDemo.
- Add an option to set VP8 encoder start bitrate
usig x-google-start-bitrate line in remote SDP.
- Allow to enabled/disable VP8 hw decoder and
encoder acceleration using appRTC settings.
BUG=4046
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7775 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/src/org/webrtc/PeerConnectionAndroidTest.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
xamples/android/res/layout/activity_fullscreen.xml
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 32ec0dd0322f2b81313d08c1998073d60678eebd |
01-Dec-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 81063831-> 81073932
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7774 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 273a414b0ec2e58fdf3b817ad8b1a02f4ce15287 |
01-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report encoded frame size in VideoSendStream.
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 2c13f659c7013d4ce9dc123708b6a2d9a9ccdb2b |
28-Nov-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7763 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/server/data_socket.cc
xamples/peerconnection/server/data_socket.h
|
| 3e9ad26112c5341aebf54ff8d43216acf7099063 |
27-Nov-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Refactor iOS AppRTC parsing code.
Moved parsing code to JSON categories for the relevant objects.
No longer prefer ISAC as audio codec.
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31989005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7755 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/objc/AppRTCDemo/APPRTCAppClient.h
xamples/objc/AppRTCDemo/APPRTCAppClient.m
xamples/objc/AppRTCDemo/APPRTCConnectionManager.m
xamples/objc/AppRTCDemo/ARDSignalingParams.h
xamples/objc/AppRTCDemo/ARDSignalingParams.m
xamples/objc/AppRTCDemo/ARDUtilities.h
xamples/objc/AppRTCDemo/ARDUtilities.m
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.h
xamples/objc/AppRTCDemo/RTCICECandidate+JSON.m
xamples/objc/AppRTCDemo/RTCICEServer+JSON.h
xamples/objc/AppRTCDemo/RTCICEServer+JSON.m
xamples/objc/AppRTCDemo/RTCMediaConstraints+JSON.h
xamples/objc/AppRTCDemo/RTCMediaConstraints+JSON.m
xamples/objc/AppRTCDemo/RTCSessionDescription+JSON.h
xamples/objc/AppRTCDemo/RTCSessionDescription+JSON.m
ibjingle_examples.gyp
|
| a71bb6033b54276fe9d199508b080d47d441645b |
26-Nov-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Revert 7750 "Don't reset sequence number for a stream on deactiv..."
> Don't reset sequence number for a stream on deactivate/reactivate.
>
> BUG=chromium:431908
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/32199004
TBR=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7752 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 31f7a0e7105a50ca4b00f8c4360f0da1c1957849 |
26-Nov-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Don't reset sequence number for a stream on deactivate/reactivate.
BUG=chromium:431908
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7750 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 2faf7eea6ff6fff472e24942c6534427cca7e56d |
26-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection.""
This reverts commit 308e7ff61327d64ba5c7761ce6b58cd1fbc4847e.
Original cl description:
This adds an Android apk for running tests on the Java layer of PeerConnection.
The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7748 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/AndroidManifest.xml
pp/webrtc/androidtests/ant.properties
pp/webrtc/androidtests/build.xml
pp/webrtc/androidtests/jni/Android.mk
pp/webrtc/androidtests/project.properties
pp/webrtc/androidtests/res/drawable-hdpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-ldpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-mdpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-xhdpi/ic_launcher.png
pp/webrtc/androidtests/res/values/strings.xml
pp/webrtc/androidtests/src/org/webrtc/PeerConnectionAndroidTest.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
ibjingle_tests.gyp
|
| 58edb83fd47ed5e1fb51f6adf3e92a54da3916db |
26-Nov-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Add video encoder fps and bitrate statistics to
Android AppRTCDemo UI.
BUG=4045
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7747 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/layout/activity_fullscreen.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 008731868a09e2fe01da53733a612dc24761f791 |
25-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement settable min/start/max bitrates in Call.
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/26199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| dab5d92df6e195b016f3a2e238f8e7a1cd5f9097 |
24-Nov-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Use mirror image for Android AppRTCDemo local preview.
Similar to Chrome apprtc using mirror image for camera
local preview provides better experience when device
is rotated.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7741 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 8562f23acb15f555f35f0b5760adc8bd1b988406 |
24-Nov-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
OWNERS: Remove tomasl@ and mallinath@
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| 308e7ff61327d64ba5c7761ce6b58cd1fbc4847e |
23-Nov-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."
This reverts r7732
Reason: Breaks compilation on Linux:
[813/818] LINK libjingle_media_unittest
FAILED: cd ../../talk; build/build_jar.sh /usr/lib/jvm/java-7-openjdk-amd64 ../out/Debug/libjingle_peerconnection_test.jar ../out/Debug/obj/talk/libjingle_peerconnection_test_jar.gen app/webrtc/javatests/src:../out/Debug/libjingle_peerconnection.jar:../third_party/junit/junit-4.11.jar app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
build/build_jar.sh: Entering directory `/mnt/data/b/build/slave/linux64/build/src/talk'
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46:warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;
^
app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java:36:error: cannot find symbol
@Test
^
symbol: class Test
location: class PeerConnectionTestJava
app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java:43:error: cannot find symbol
@Test
^
symbol: class Test
location: class PeerConnectionTestJava
2 errors
1 warning
ninja: build stopped: subcommand failed.
TBR=perkj@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/32169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7733 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/AndroidManifest.xml
pp/webrtc/androidtests/ant.properties
pp/webrtc/androidtests/build.xml
pp/webrtc/androidtests/jni/Android.mk
pp/webrtc/androidtests/project.properties
pp/webrtc/androidtests/res/drawable-hdpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-ldpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-mdpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-xhdpi/ic_launcher.png
pp/webrtc/androidtests/res/values/strings.xml
pp/webrtc/androidtests/src/org/webrtc/PeerConnectionAndroidTest.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
ibjingle_tests.gyp
|
| 2751f2ab4c6aa7eebf4bdac1ba72ea41d3975adf |
23-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
This adds an Android apk for running tests on the Java layer of PeerConnection.
The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner
R=kjellander@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7732 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/androidtests/AndroidManifest.xml
pp/webrtc/androidtests/ant.properties
pp/webrtc/androidtests/build.xml
pp/webrtc/androidtests/jni/Android.mk
pp/webrtc/androidtests/project.properties
pp/webrtc/androidtests/res/drawable-hdpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-ldpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-mdpi/ic_launcher.png
pp/webrtc/androidtests/res/drawable-xhdpi/ic_launcher.png
pp/webrtc/androidtests/res/values/strings.xml
pp/webrtc/androidtests/src/org/webrtc/PeerConnectionAndroidTest.java
pp/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
ibjingle_tests.gyp
|
| 88d14f483b43f5c34c258607dd127bc64308c927 |
22-Nov-2014 |
thorcarpenter@google.com <thorcarpenter@google.com> |
Remove expensive and unnecessary memory alloc for sending black frames on video
mute.
Remove old crusty is_black_ member var in webrtcvideoengine which was not adding value.
R=henrike@webrtc.org, tpsiaki@google.com
Review URL: https://webrtc-codereview.appspot.com/26229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7731 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| bdcf38c89446b1b464a646414f6cd7573a190bd1 |
21-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class
There is also an implementation in Chromium that can be removed if/when this lands:
content/renderer/media/webrtc/webrtc_video_capturer_adapter.cc
R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7728 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
edia/base/videoframe.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoframe.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| edc6e57a92d2b366871f4c2d2e926748326017b9 |
20-Nov-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Support loopback mode and command line execution
for Android AppRTCDemo when using WebSocket signaling.
- Add loopback support for new signaling. In loopback mode
only room connection is established, WebSocket connection is
not opened and all candidate/sdp messages are automatically
routed back.
- Fix command line support both for channek and new signaling.
Exit from application when room connection is closed and add
an option to run application for certain time period and exit.
- Plus some fixes for WebSocket signaling - support
POST (not used for now) and DELETE request to WebSocket server
and making sure that all available TURN server are used by
peer connection client.
BUG=3995,3937
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7725 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/GAERTCClient.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/third_party/autobanh/NOTICE
|
| f58b455cf7e8a3076c216f856a1e8a93c3c4d31c |
19-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7702
Committed: https://code.google.com/p/webrtc/source/detail?r=7707
Review URL: https://webrtc-codereview.appspot.com/29949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7721 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videoframefactory.cc
edia/base/videoframefactory.h
|
| 6f6ef72950b9bda79392e83d7b1495d4ff07b4a2 |
19-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add DCHECK to ensure that NetEq's packet buffer is not empty
This DCHECK ensures that one packet was inserted after the buffer was
flushed.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/GAEChannelClient.java
xamples/android/src/org/appspot/apprtc/GAERTCClient.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/RoomParametersFetcher.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/WebSocketChannelClient.java
xamples/android/src/org/appspot/apprtc/WebSocketRTCClient.java
xamples/android/third_party/autobanh/LICENSE
xamples/android/third_party/autobanh/LICENSE.md
xamples/android/third_party/autobanh/autobanh.jar
ibjingle_examples.gyp
|
| 2176db343cf269a6f1faa7f0b20e8b5ad001c654 |
18-Nov-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land)
This CL was incorrectly reverted in r7647 by the libjingle sync bot.
TBR=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/32489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7717 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 930e004a817ed346a99ac8e56575326ca75e72aa |
17-Nov-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.
BUG=427555
Copied from https://webrtc-codereview.appspot.com/25959004/
R=harryjin@google.com, juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7693
Review URL: https://webrtc-codereview.appspot.com/32039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
|
| c72a22c23d1174a58976e61bce809fd7d2e71399 |
17-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
Add preliminary empty file videoframefactory.cc
The purpose of this CL is to add a new file in libjingle without breaking Chromium in the process. The plan is to do the following:
1. Land a no-op videoframefactory.cc in webrtc (this file).
2. Wait for it to roll into Chromium.
3. Modify libjingle.gyp in Chromium to include this file.
4. Make the real change in webrtc with the real implementation of this file.
5. Wait for the change to roll into Chromium.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7712 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframefactory.cc
|
| 4ef22d1d293fe7b2398e4cd90a0eb2e8fb02b6ea |
17-Nov-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Setting Opus FEC as default
BUG=3986
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7710 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 4ec19e306aa4424c7d93763ef11ff150552d849f |
16-Nov-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..."
This didn't compile on the FYI bots. Example error:
FAILED: E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-with-manifests environment.x86 True chrome_child.dll "E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-wrapper environment.x86 False link.exe /nologo /IMPLIB:chrome_child.dll.lib /DLL /OUT:chrome_child.dll @chrome_child.dll.rsp" 2 mt.exe rc.exe "obj\chrome\chrome_child_dll.chrome_child.dll.intermediate.manifest" obj\chrome\chrome_child_dll.chrome_child.dll.generated.manifest
content_renderer.lib(content_renderer.webrtc_video_capturer_adapter.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
libjingle_webrtc_common.lib(libjingle_webrtc_common.peerconnectionfactory.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
libjingle_webrtc_common.lib(libjingle_webrtc_common.videocapturer.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
libjingle_webrtc_common.lib(libjingle_webrtc_common.dummydevicemanager.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)
chrome_child.dll : fatal error LNK1120: 1 unresolved externals
> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
>
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
>
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
>
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
>
> Committed: https://code.google.com/p/webrtc/source/detail?r=7702
>
> Review URL: https://webrtc-codereview.appspot.com/29949004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7708 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videoframefactory.cc
edia/base/videoframefactory.h
|
| 858dbbced270a09582aa4bd374f06b506363bd4d |
16-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7702
Review URL: https://webrtc-codereview.appspot.com/29949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7707 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videoframefactory.cc
edia/base/videoframefactory.h
|
| 6a782c2a46d83e09bb036d34b8c2363adc26d037 |
14-Nov-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
TBR=guoweis@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/25179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
|
| a73d746562a556c14905939f4d779038b1d5cb8e |
14-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..."
Rease for revert: failed internal test cases
> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
>
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
>
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
>
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/29949004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7703 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videoframefactory.cc
edia/base/videoframefactory.h
|
| bbd8cad21f276e1b5603ed038d18b62bc18a2de7 |
14-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7702 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videoframefactory.cc
edia/base/videoframefactory.h
|
| ece3890d3a40fe911ae895e28c329491e795b14d |
14-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report total bitrate for all streams in GetStats.
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.
R=stefan@webrtc.org, xians@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/27179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
|
| 35c1ace18532b50ff274f65b1369889baefca319 |
13-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..."
Reason for revert is failed testcases:
WebRtcVideoEngineExtendedTestFake.ResetSimulcastSendCodecOnNewFrameSize
WebRtcVideoEngineExtendedTestFake.MultipleSendStreamsDifferentFormats
> WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
>
> BUG=3936
> R=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/30039004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7700 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideocapturer.h
edia/base/videocapturer_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
ession/media/channelmanager_unittest.cc
|
| a1f5b96351d0aa9ea42f768a32c9f717497dd427 |
13-Nov-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove unnecessary copying of libjingle resource files.
This copying has probably not been needed since
https://code.google.com/p/webrtc/source/detail?r=7088
BUG=398
TESTED=Removed the top-level talk directory and ran
libjingle_media_unittest from the following working directories:
* checkout-root/src/out/Debug
* checkout-root/src
* checkout-root
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7699 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 52da44b7e61bb6cf3f36c8c4f8de1a888e2814bb |
13-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
BUG=3936
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7698 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakevideocapturer.h
edia/base/videocapturer_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
ession/media/channelmanager_unittest.cc
|
| 312614a438c2104ccab6d0231d17604359674e15 |
13-Nov-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add jmi field for packets discarded due to network error
Also included the total packets attempted to send.
BUG=427555
Copied from https://webrtc-codereview.appspot.com/25959004/
R=harryjin@google.com, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
|
| 6ca6190be2bdb6dc50a77e393e747cf8f55ea2c0 |
12-Nov-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix a SCTP message reordering issue in datachannel.cc.
Previously DataChannel::SendQueuedDataMessages continues the loop of sending queued messages if the channel is blocked, which will cause message reordering if the channel becomes unblocked during the loop, i.e. messages attempted after the unblocking will be sent earlier than the older messages attempted before the unblocking.
BUG=3979
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7690 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
|
| 8038d42749e9edd52487baea050acda6f604bf91 |
11-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Follow-up fixes for G722
This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.
BUG=3951
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| c4922316b417ebe35ab006a2945de889a26b9d4e |
10-Nov-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds.
TBR=niklas.enbom@webrtc.org
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/30959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7670 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/basicpacketsocketfactory.cc
2p/base/basicpacketsocketfactory.h
2p/base/candidate.h
2p/base/common.h
2p/base/constants.cc
2p/base/constants.h
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransport.cc
2p/base/p2ptransport.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/packetsocketfactory.h
2p/base/parsing.cc
2p/base/parsing.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portallocator.cc
2p/base/portallocator.h
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy.h
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/pseudotcp.cc
2p/base/pseudotcp.h
2p/base/pseudotcp_unittest.cc
2p/base/rawtransport.cc
2p/base/rawtransport.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/relayserver_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/sessionclient.h
2p/base/sessiondescription.cc
2p/base/sessiondescription.h
2p/base/sessionid.h
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/stun.cc
2p/base/stun.h
2p/base/stun_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunrequest.cc
2p/base/stunrequest.h
2p/base/stunrequest_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/stunserver_unittest.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/testrelayserver.h
2p/base/teststunserver.h
2p/base/testturnserver.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.cc
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
2p/base/transportdescriptionfactory_unittest.cc
2p/base/transportinfo.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/base/udpport.h
2p/client/autoportallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
2p/client/httpportallocator.cc
2p/client/httpportallocator.h
2p/client/portallocator_unittest.cc
2p/client/sessionmanagertask.h
2p/client/sessionsendtask.h
2p/client/socketmonitor.cc
2p/client/socketmonitor.h
mllite/qname.cc
mllite/qname.h
mllite/qname_unittest.cc
mllite/xmlbuilder.cc
mllite/xmlbuilder.h
mllite/xmlbuilder_unittest.cc
mllite/xmlconstants.cc
mllite/xmlconstants.h
mllite/xmlelement.cc
mllite/xmlelement.h
mllite/xmlelement_unittest.cc
mllite/xmlnsstack.cc
mllite/xmlnsstack.h
mllite/xmlnsstack_unittest.cc
mllite/xmlparser.cc
mllite/xmlparser.h
mllite/xmlparser_unittest.cc
mllite/xmlprinter.cc
mllite/xmlprinter.h
mllite/xmlprinter_unittest.cc
mpp/asyncsocket.h
mpp/chatroommodule.h
mpp/chatroommodule_unittest.cc
mpp/chatroommoduleimpl.cc
mpp/constants.cc
mpp/constants.h
mpp/discoitemsquerytask.cc
mpp/discoitemsquerytask.h
mpp/fakexmppclient.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/hangoutpubsubclient_unittest.cc
mpp/iqtask.cc
mpp/iqtask.h
mpp/jid.cc
mpp/jid.h
mpp/jid_unittest.cc
mpp/jingleinfotask.cc
mpp/jingleinfotask.h
mpp/module.h
mpp/moduleimpl.cc
mpp/moduleimpl.h
mpp/mucroomconfigtask.cc
mpp/mucroomconfigtask.h
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask.cc
mpp/mucroomdiscoverytask.h
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.cc
mpp/mucroomlookuptask.h
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask.cc
mpp/mucroomuniquehangoutidtask.h
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask.cc
mpp/pingtask.h
mpp/pingtask_unittest.cc
mpp/plainsaslhandler.h
mpp/presenceouttask.cc
mpp/presenceouttask.h
mpp/presencereceivetask.cc
mpp/presencereceivetask.h
mpp/presencestatus.cc
mpp/presencestatus.h
mpp/prexmppauth.h
mpp/pubsub_task.cc
mpp/pubsub_task.h
mpp/pubsubclient.cc
mpp/pubsubclient.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks.cc
mpp/pubsubtasks.h
mpp/pubsubtasks_unittest.cc
mpp/receivetask.cc
mpp/receivetask.h
mpp/rostermodule.h
mpp/rostermodule_unittest.cc
mpp/rostermoduleimpl.cc
mpp/rostermoduleimpl.h
mpp/saslcookiemechanism.h
mpp/saslhandler.h
mpp/saslmechanism.cc
mpp/saslmechanism.h
mpp/saslplainmechanism.h
mpp/util_unittest.cc
mpp/util_unittest.h
mpp/xmppauth.cc
mpp/xmppauth.h
mpp/xmppclient.cc
mpp/xmppclient.h
mpp/xmppclientsettings.h
mpp/xmppengine.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmppengineimpl.h
mpp/xmppengineimpl_iq.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask.h
mpp/xmpplogintask_unittest.cc
mpp/xmpppump.cc
mpp/xmpppump.h
mpp/xmppsocket.cc
mpp/xmppsocket.h
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser.h
mpp/xmppstanzaparser_unittest.cc
mpp/xmpptask.cc
mpp/xmpptask.h
mpp/xmppthread.cc
mpp/xmppthread.h
|
| d819803d4570564a9800a7dd54f4593e6e21a6e7 |
10-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up DSCP support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/24249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 957e802fe0e6e765425955cc1e3e02f73d1a670b |
10-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor SetDefaultEncoderConfig to work on existing codecs.
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/32009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 3c1970f9f38d042e8eadb1a6bee74d99a2781e65 |
07-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79414100-> 79428003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7664 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/connectivitychecker.cc
|
| 188d3b2245b49f21468840386d81b080176b434b |
07-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Enable VP9 video codec support on webrtcvideoengine behind a field trial.
BUG=chromium:431285
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7663 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/constants.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| f85dbce041a9c49252b5c27364ce70300b652d78 |
07-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.
BUG=3951
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/mediasessionclient_unittest.cc
|
| d105cc81dc1f5792fd4165d6aec0a654f2dfc77c |
07-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Change dummy address to use 0.0.0.0 instead of ::
This is to not break compatiblity with FF.
https://code.google.com/p/chromium/issues/detail?id=430333
TBR=pthatcher@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7661 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| a2ef4fe9c331e7668b9e8ff64ce5141a535a5f21 |
07-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Prevent a lot of VideoSendStream reconfigures.
Checking whether we're setting the same configuration or not.
Experimentally this brings down underlying reconfigures from ~20 to
about 4-5.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 82775b13965b4d41299b097c09c30c4ab160cdac |
07-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime.
This will allow to plugin VP9 based on a field trial.
R=pbos@webrtc.org, pbos, pthatcher
Review URL: https://webrtc-codereview.appspot.com/27949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/constants.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
|
| 5e160660a64e5bb1afd3ae546bc147f8c0a893c5 |
06-Nov-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
Reland Volume buttons in AppRTCDemo should affect output audio volume (part I).
Second attempt to land https://webrtc-codereview.appspot.com/32399004/
TBR=perkj@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7657 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
ibjingle_examples.gyp
|
| dced5d7835ec8ada6242c2086af7899f068e96ed |
06-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert "Advertise G722 as 8 kHz rather than 16 kHz"
This reverts r7645.
TBR=pthatcher@webrtc.org
BUG=3951
Review URL: https://webrtc-codereview.appspot.com/24199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 34bda43fa63e2f1bab7272700cea208931ee2d85 |
06-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79326895-> 79329222
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
ibjingle_examples.gyp
|
| e5421e9602d920fc419245ff795ee29b7eb758c5 |
06-Nov-2014 |
henrika@webrtc.org <henrika@webrtc.org> |
Volume buttons in AppRTCDemo should affect output audio volume.
BUG=3279
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCAudioManager.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
ibjingle_examples.gyp
|
| fd0efb694acef8f592872496e1b443b0e2271e74 |
06-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Remove deprecated PeerConnection APIs.
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.
R=tommi@chromium.org
Review URL: https://webrtc-codereview.appspot.com/24189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
|
| 19b47410044aecac20f3f46a4d207018fc466e2e |
06-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Removing unused method GetDefaultVideoEncoderConfig.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 0ef890a4babc995f1a8bff2a13be972a162bca70 |
06-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79285346-> 79320771
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 6340acde6887226638c4ac2662fb54a529a9d93f |
06-Nov-2014 |
mcasas@webrtc.org <mcasas@webrtc.org> |
AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation.
Also removed some unused "summary" ListPreference
fields.
The looks of it can be found in [1] (lowest row).
[1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
|
| 1dcca4028fe06735819ec1ba89e5814d53767a4b |
06-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Advertise G722 as 8 kHz rather than 16 kHz
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.
R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.
Review URL: https://webrtc-codereview.appspot.com/27879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| ee9d61ce4547d1704a70548890e7c447d14bbe7e |
05-Nov-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
This fixes a small memory leak (found using Xcode/Instruments on iOS) in
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient
BUG=3985
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28959004
Patch from Matthias Liebig <matthias.gcode@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnection.mm
|
| 0bae1fab4adb9bb8164e53142bf419049eafec38 |
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| a22a628356920c6a4cc785bf77077f48d55ee8ac |
05-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79205306-> 79244016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7633 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 795d00377078305cc468530bbb5d2d782c27c792 |
05-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79200114-> 79205306
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 8125744a5f95101d5f3ce376c1248727c68b142f |
05-Nov-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Cleanup RTCVideoRenderer interface.
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.
BUG=3795
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCEAGLVideoView+Internal.h
pp/webrtc/objc/RTCEAGLVideoView.m
pp/webrtc/objc/RTCI420Frame.mm
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCNSGLVideoView.m
pp/webrtc/objc/RTCVideoRenderer+Internal.h
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/RTCVideoRendererAdapter.h
pp/webrtc/objc/RTCVideoRendererAdapter.mm
pp/webrtc/objc/RTCVideoTrack+Internal.h
pp/webrtc/objc/RTCVideoTrack.mm
pp/webrtc/objc/public/RTCEAGLVideoView.h
pp/webrtc/objc/public/RTCI420Frame.h
pp/webrtc/objc/public/RTCNSGLVideoView.h
pp/webrtc/objc/public/RTCVideoRenderer.h
pp/webrtc/objc/public/RTCVideoTrack.h
pp/webrtc/objctests/RTCPeerConnectionTest.mm
xamples/objc/AppRTCDemo/ios/APPRTCViewController.m
xamples/objc/AppRTCDemo/mac/APPRTCViewController.m
ibjingle.gyp
|
| 45ecf4c092ad15fff70e8d5382de3c3d0cfe4aba |
04-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79169148-> 79192489
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.h
|
| 8944c9d08b6585b25161ecafc60014e158ef3a25 |
04-Nov-2014 |
mcasas@webrtc.org <mcasas@webrtc.org> |
AppRTCDemoActivity: use differnet Themes for different API levels
The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.
I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)
Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/AndroidManifest.xml
xamples/android/res/values-v21/styles.xml
xamples/android/res/values/styles.xml
|
| fad9aecbf5c461ee5f1ad590b52d010cf9f5afa3 |
04-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove protected files from talk/PRESUBMIT.py.
All files may now be committed to.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| 88ef6322864b4071df4ed724a3989a9183d92172 |
04-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Falling back on single-stream on multiple SSRC.
Instead of failing, use one stream. Also clamp video min bitrate.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| b5d045e94df5e434637d2fbaf68feb95adb1b541 |
04-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
|
| 18de6f96226110206b015be961849381340d7d7c |
04-Nov-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send.
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.
See e.g. crbug.com/429740 (and more)
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/proxy.h
|
| c2dd5ee2c05b466949fedae3fcfac63838104392 |
04-Nov-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.
OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/objc/public/RTCPeerConnectionDelegate.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/objc/AppRTCDemo/APPRTCConnectionManager.m
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
|
| a663d90ae3cc70da5205e6d4f0924b3236916122 |
03-Nov-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 79104430-> 79104922
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 5f38c8d1b8e842652e55410333870acfc5395ea6 |
03-Nov-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Android AppRTCDemo improvements:
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.
BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/res/drawable-hdpi/ic_action_full_screen.png
xamples/android/res/drawable-hdpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-hdpi/ic_loopback_call.png
xamples/android/res/drawable-ldpi/ic_action_full_screen.png
xamples/android/res/drawable-ldpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-ldpi/ic_loopback_call.png
xamples/android/res/drawable-mdpi/ic_action_full_screen.png
xamples/android/res/drawable-mdpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-mdpi/ic_loopback_call.png
xamples/android/res/drawable-xhdpi/ic_action_full_screen.png
xamples/android/res/drawable-xhdpi/ic_action_return_from_full_screen.png
xamples/android/res/drawable-xhdpi/ic_loopback_call.png
xamples/android/res/layout/activity_connect.xml
xamples/android/res/layout/fragment_menubar.xml
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/GAEChannelClient.java
xamples/android/src/org/appspot/apprtc/GAERTCClient.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
ibjingle_examples.gyp
|
| 96a93259b361f4b03080a188d781b0835cf4edaf |
03-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement external decoder support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 2236267b5ef06bf16dd3d09df094103ac502260f |
03-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan
This test is flaky on MSan bots.
BUG=3980
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| 5072e0f6cd27bc8f8c0788644c514f344d5ac3f6 |
01-Nov-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Update Android projects to API level 21.
The update in https://webrtc-codereview.appspot.com/23309004
was not enough, so this updates to 21 instead.
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 20.
Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.
BUG=
R=glaznev@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/project.properties
|
| c2c94a9a9f8e7c58c8ac7d228090c0eff76b282c |
31-Oct-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64
Given that OpenJDK 1.7 is the recommended Java SDK for
Chromium these days, we should get rid of linking to the old
non-standardized link referring to a Sun Java 1.6 SDK.
Instead of requiring all users to set JAVA_HOME, I prefer
have the most common path as default and and close webrtc:2113
as won't fix after this is submitted.
BUG=2113
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| 78c222bfae49603532be53097cca10225738b79c |
31-Oct-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Update all .isolate files for the new format.
R=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27809004
Patch from Marc-Antoine Ruel <maruel@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_tests.gyp
ibjingle_unittest.isolate
|
| 8a130c1084b2b76f4511ca6eae6468da43022ce5 |
31-Oct-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Update Android projects to API level 20.
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 19.
Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.
BUG=
R=glaznev@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/project.properties
|
| b7ed7799e77d3b315f5016951ecb90d18f10fdcb |
31-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement conference-mode temporal-layer screencast.
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 3bf3d238c8c4578e444e5a601684db68c79a29ca |
31-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Configure A/V sync in WebRtcVideoEngine2.
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/23249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 2dc6f3154dd233b221c53272a7f64aa20ef2e95e |
31-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Adapting bitrate according to maxplaybackrate for Opus.
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 14146e40aaafadbd7fb99ea9e45f528624f71538 |
31-Oct-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
arm64 iOS build.
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.
BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
|
| 50ca986bc19e5a6ee7e7f6265dab8558bea979a3 |
31-Oct-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Improve the logging when a TCP connection is deleted.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port.cc
2p/base/tcpport.cc
|
| 8219529b98238244ed4b57acaff4e0b9bf9ddca4 |
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Cleaning up r7562-7567.
Wrongly used git svn dcommit for committing a CL.
Then two reverts were applied.
Still something needs to be cleaned.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| 879fac81d15cca19f1c9edf48833ac27637fe536 |
30-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78822708-> 78823675
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| 5f73a375973a8917f6d417aa7d2d2fe80856b6b0 |
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Revert 7563 "before rebase" due to wrong submission
> before rebase
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| c11cc8d9475a35d559c8127cba0fa22478d6e36d |
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Revert 7564 "to submit" due to wrong submission
> to submit
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| de386bf67b76e48b9c0c58580938b91b644f42f8 |
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
to submit
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| c673bb9f29fb0c80c112b91942682475560f821d |
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
before rebase
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| 0b626725761cd89d4422f4538939613cbe5d1f27 |
30-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
adding default rates
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| 776e6f289c7396a1143b8b36b03f88b08ac8cba3 |
29-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Use external VideoDecoders in VideoReceiveStream.
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 1abc146aa5e474471f866a8a4db7dcfa8fb7c8a6 |
29-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78738075-> 78738103
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
|
| 79980897890aa0563024a58ccd580694ed747804 |
29-Oct-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
ApprtDemo Android: Switch between front and back camera.
This adds a UI icon for switching between the front and back camera.
This cl adds the possibility to change between the front and back camera while in a call
or before the other end have connected.
BUG=3786
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7553 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/res/layout/fragment_menubar.xml
xamples/android/res/values/strings.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
|
| 2623695dfb48ebd745d0d578f5720e8d5160f4f3 |
29-Oct-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Renaming bandwidth to bitrate in webrtcvoiceengine.
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.
This is to remove the confusion inside webrtcvoiceengine
BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
xamples/call/call_main.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/friendinvitesendtask.cc
xamples/call/friendinvitesendtask.h
xamples/call/muc.h
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.cc
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
edia/base/fakemediaengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/basicpacketsocketfactory.cc
2p/base/basicpacketsocketfactory.h
2p/base/candidate.h
2p/base/common.h
2p/base/constants.cc
2p/base/constants.h
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransport.cc
2p/base/p2ptransport.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/packetsocketfactory.h
2p/base/parsing.cc
2p/base/parsing.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portallocator.cc
2p/base/portallocator.h
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy.h
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/pseudotcp.cc
2p/base/pseudotcp.h
2p/base/pseudotcp_unittest.cc
2p/base/rawtransport.cc
2p/base/rawtransport.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/relayserver_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/sessionclient.h
2p/base/sessiondescription.cc
2p/base/sessiondescription.h
2p/base/sessionid.h
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/stun.cc
2p/base/stun.h
2p/base/stun_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunrequest.cc
2p/base/stunrequest.h
2p/base/stunrequest_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/stunserver_unittest.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/testrelayserver.h
2p/base/teststunserver.h
2p/base/testturnserver.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.cc
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
2p/base/transportdescriptionfactory_unittest.cc
2p/base/transportinfo.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/base/udpport.h
2p/client/autoportallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
2p/client/httpportallocator.cc
2p/client/httpportallocator.h
2p/client/portallocator_unittest.cc
2p/client/sessionmanagertask.h
2p/client/sessionsendtask.h
2p/client/socketmonitor.cc
2p/client/socketmonitor.h
ession/media/audiomonitor.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediarecorder_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/media/rtcpmuxfilter.h
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
ession/media/typingmonitor_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
mpp/chatroommodule.h
mpp/chatroommoduleimpl.cc
mpp/constants.cc
mpp/constants.h
mpp/discoitemsquerytask.cc
mpp/discoitemsquerytask.h
mpp/fakexmppclient.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/hangoutpubsubclient_unittest.cc
mpp/iqtask.cc
mpp/iqtask.h
mpp/jid.cc
mpp/jid_unittest.cc
mpp/jingleinfotask.cc
mpp/jingleinfotask.h
mpp/module.h
mpp/moduleimpl.cc
mpp/moduleimpl.h
mpp/mucroomconfigtask.cc
mpp/mucroomconfigtask.h
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask.cc
mpp/mucroomdiscoverytask.h
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.cc
mpp/mucroomlookuptask.h
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask.cc
mpp/mucroomuniquehangoutidtask.h
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask.cc
mpp/pingtask.h
mpp/pingtask_unittest.cc
mpp/plainsaslhandler.h
mpp/presenceouttask.cc
mpp/presenceouttask.h
mpp/presencereceivetask.cc
mpp/presencereceivetask.h
mpp/presencestatus.cc
mpp/presencestatus.h
mpp/prexmppauth.h
mpp/pubsub_task.cc
mpp/pubsub_task.h
mpp/pubsubclient.cc
mpp/pubsubclient.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks.cc
mpp/pubsubtasks.h
mpp/pubsubtasks_unittest.cc
mpp/receivetask.cc
mpp/receivetask.h
mpp/rostermodule.h
mpp/rostermodule_unittest.cc
mpp/rostermoduleimpl.cc
mpp/rostermoduleimpl.h
mpp/saslcookiemechanism.h
mpp/saslmechanism.cc
mpp/saslplainmechanism.h
mpp/util_unittest.cc
mpp/util_unittest.h
mpp/xmppauth.cc
mpp/xmppauth.h
mpp/xmppclient.cc
mpp/xmppclient.h
mpp/xmppclientsettings.h
mpp/xmppengine.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmppengineimpl.h
mpp/xmppengineimpl_iq.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask.h
mpp/xmpplogintask_unittest.cc
mpp/xmpppump.cc
mpp/xmpppump.h
mpp/xmppsocket.h
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser_unittest.cc
mpp/xmpptask.cc
mpp/xmpptask.h
mpp/xmppthread.cc
mpp/xmppthread.h
|
| ae694effd85d501f15600275dec96522a00c4feb |
28-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78642371-> 78680406
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| fbd55cb27db1b84b20e8884da348c7b7d957281e |
28-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78616359-> 78642371
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7540 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| f15dee6980152cded2f10c26748d7d88ab9501ae |
27-Oct-2014 |
tommi@webrtc.org <tommi@webrtc.org> |
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 243eb8e9af35f07befa733c86dd320f9f8b021bd |
27-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Adding setting screen to AppRTCDemo.
- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.
BUG=3935,3953
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/AndroidManifest.xml
xamples/android/res/layout/activity_connect.xml
xamples/android/res/menu/connect_menu.xml
xamples/android/res/values/arrays.xml
xamples/android/res/values/strings.xml
xamples/android/res/xml/preferences.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
xamples/android/src/org/appspot/apprtc/SettingsActivity.java
xamples/android/src/org/appspot/apprtc/SettingsFragment.java
ibjingle_examples.gyp
|
| 068b529f46c8f6033ad2ac1d182b70c4d67ffb11 |
27-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78583324-> 78583691
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7532 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 2e7ee4b28bbdf92bdf804b600ae33679d1799788 |
27-Oct-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Fix the SrtpFilter crash caused by two local offers.
BUG=http://crbug.com/421774
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/srtpfilter.cc
ession/media/srtpfilter_unittest.cc
|
| efc82c2c734171faba9e400ff60a114e7af2ebcc |
27-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement screencast settings for WebRtcVideoEngine2.
Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 1732df61294d85fc8e4452985e775099e150afe4 |
27-Oct-2014 |
braveyao@webrtc.org <braveyao@webrtc.org> |
Use flags set by the port allocator.
Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.
BUG=webrtc:3958
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 3f7bcc126d85e82f60e3dc4135562e1a704b1a83 |
24-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78430441-> 78445452
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7522 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| c7ed8db7fd29f09f202d69b5191bf7e954fc6916 |
24-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78427027-> 78430441
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7521 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 470988742a2ad2edbeba3a99b86481b4fb0cd0d3 |
24-Oct-2014 |
perkj@webrtc.org <perkj@webrtc.org> |
Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.
BUG=3934
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
|
| c9d6d140209bd2e8f44eb41fb0de17d512d39911 |
24-Oct-2014 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
patch from issue 25469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 8fe75ee2344ac3106027b2c7d150ab0bd4165ef8 |
24-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78381351-> 78389679
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7516 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| fb5e9fc44e04a4e414c429e22ac6b42c1bfce62a |
23-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78344087-> 78381351
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 580d367b1482b2472f6c220a5c30d3942524f36c |
23-Oct-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add macros and APIs for webrtc histograms.
BUG=crbug/419657
Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.
R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 9d446f2e167d0697364a118a3217ddaa47a3ce4d |
23-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78296920-> 78342456
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| a9f0898e7dc7ba89d3ba8c3a2c2c4a32c79a36ed |
23-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78273470-> 78296920
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 7bb4a9881df1cd8b26391d9f15ef31117396ff19 |
22-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Merging Henrik's and Peter's changes for AppRTCDemo
from https://github.com/hkjellander/AppRTCDemo.
Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.
BUG=3938
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/AndroidManifest.xml
xamples/android/res/drawable-hdpi/disconnect.png
xamples/android/res/drawable-hdpi/ic_launcher.png
xamples/android/res/drawable-ldpi/disconnect.png
xamples/android/res/drawable-mdpi/disconnect.png
xamples/android/res/drawable-mdpi/ic_launcher.png
xamples/android/res/drawable-xhdpi/disconnect.png
xamples/android/res/drawable-xhdpi/ic_launcher.png
xamples/android/res/layout/activity_connect.xml
xamples/android/res/layout/activity_fullscreen.xml
xamples/android/res/layout/fragment_menubar.xml
xamples/android/res/values/strings.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/AppRTCGLView.java
xamples/android/src/org/appspot/apprtc/ConnectActivity.java
ibjingle_examples.gyp
|
| fb5410a8b7fbceb7c80c17324d072ece09961a1d |
22-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78262388-> 78262615
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| eacc6e4657be1f2a0d5a182130ea04f85709a2f5 |
22-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove some disabled tests in WebRtcVideoEngine2.
Removes some tests that shouldn't have to be implemented or have already
been through other tests.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7495 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| a5c36b397a099e2c91b6ee7f6951249b3fec9ffa |
21-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78193292-> 78199328
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| b6173abe59e11b749295482aa7f30c8fc3d2f47e |
21-Oct-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Fix local address leakage when IceTransportsType is relay
As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage.
BUG=1179
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port.cc
2p/base/port.h
2p/base/stunport.cc
2p/base/turnport.cc
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
|
| 1288cbb7046da92857d423f2fe796826c76a04da |
21-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78106439-> 78193292
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7482 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| a8c0edd29f440f933b3870ccd4f50003fad6e6a5 |
20-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Avoid using EGLContext class for Android 4.1 and below.
Support for this class was added in Android 4.2, so
disable surface decoding for lower Android versions.
BUG=3901
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7478 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| fa553ef6053b20f3768d5fe4314e8c993648bf0a |
20-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set up start bitrate in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/27789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
xamples/call/call_main.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/friendinvitesendtask.cc
xamples/call/friendinvitesendtask.h
xamples/call/muc.h
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.cc
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
edia/base/fakemediaengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/basicpacketsocketfactory.cc
2p/base/basicpacketsocketfactory.h
2p/base/candidate.h
2p/base/common.h
2p/base/constants.cc
2p/base/constants.h
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransport.cc
2p/base/p2ptransport.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/packetsocketfactory.h
2p/base/parsing.cc
2p/base/parsing.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portallocator.cc
2p/base/portallocator.h
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy.h
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/pseudotcp.cc
2p/base/pseudotcp.h
2p/base/pseudotcp_unittest.cc
2p/base/rawtransport.cc
2p/base/rawtransport.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/relayserver_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/sessionclient.h
2p/base/sessiondescription.cc
2p/base/sessiondescription.h
2p/base/sessionid.h
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/stun.cc
2p/base/stun.h
2p/base/stun_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunrequest.cc
2p/base/stunrequest.h
2p/base/stunrequest_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/stunserver_unittest.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/testrelayserver.h
2p/base/teststunserver.h
2p/base/testturnserver.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.cc
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
2p/base/transportdescriptionfactory_unittest.cc
2p/base/transportinfo.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/base/udpport.h
2p/client/autoportallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
2p/client/httpportallocator.cc
2p/client/httpportallocator.h
2p/client/portallocator_unittest.cc
2p/client/sessionmanagertask.h
2p/client/sessionsendtask.h
2p/client/socketmonitor.cc
2p/client/socketmonitor.h
ession/media/audiomonitor.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediarecorder_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/media/rtcpmuxfilter.h
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
ession/media/typingmonitor_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
mpp/chatroommodule.h
mpp/chatroommoduleimpl.cc
mpp/constants.cc
mpp/constants.h
mpp/discoitemsquerytask.cc
mpp/discoitemsquerytask.h
mpp/fakexmppclient.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/hangoutpubsubclient_unittest.cc
mpp/iqtask.cc
mpp/iqtask.h
mpp/jid.cc
mpp/jid_unittest.cc
mpp/jingleinfotask.cc
mpp/jingleinfotask.h
mpp/module.h
mpp/moduleimpl.cc
mpp/moduleimpl.h
mpp/mucroomconfigtask.cc
mpp/mucroomconfigtask.h
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask.cc
mpp/mucroomdiscoverytask.h
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.cc
mpp/mucroomlookuptask.h
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask.cc
mpp/mucroomuniquehangoutidtask.h
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask.cc
mpp/pingtask.h
mpp/pingtask_unittest.cc
mpp/plainsaslhandler.h
mpp/presenceouttask.cc
mpp/presenceouttask.h
mpp/presencereceivetask.cc
mpp/presencereceivetask.h
mpp/presencestatus.cc
mpp/presencestatus.h
mpp/prexmppauth.h
mpp/pubsub_task.cc
mpp/pubsub_task.h
mpp/pubsubclient.cc
mpp/pubsubclient.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks.cc
mpp/pubsubtasks.h
mpp/pubsubtasks_unittest.cc
mpp/receivetask.cc
mpp/receivetask.h
mpp/rostermodule.h
mpp/rostermodule_unittest.cc
mpp/rostermoduleimpl.cc
mpp/rostermoduleimpl.h
mpp/saslcookiemechanism.h
mpp/saslmechanism.cc
mpp/saslplainmechanism.h
mpp/util_unittest.cc
mpp/util_unittest.h
mpp/xmppauth.cc
mpp/xmppauth.h
mpp/xmppclient.cc
mpp/xmppclient.h
mpp/xmppclientsettings.h
mpp/xmppengine.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmppengineimpl.h
mpp/xmppengineimpl_iq.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask.h
mpp/xmpplogintask_unittest.cc
mpp/xmpppump.cc
mpp/xmpppump.h
mpp/xmppsocket.h
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser_unittest.cc
mpp/xmpptask.cc
mpp/xmpptask.h
mpp/xmppthread.cc
mpp/xmppthread.h
|
| 7992b409944597be058b43b506fc2a875518e82a |
17-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77953038-> 77970462
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7471 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
|
| 58202946a7184b5b90464b1fa0423e7cc95aa37c |
17-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Cleaning up Android AppRTCDemo.
- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.
BUG=
R=braveyao@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/GAEChannelClient.java
xamples/android/src/org/appspot/apprtc/GAERTCClient.java
xamples/android/src/org/appspot/apprtc/PeerConnectionClient.java
ibjingle_examples.gyp
|
| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
xamples/call/call_main.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/friendinvitesendtask.cc
xamples/call/friendinvitesendtask.h
xamples/call/muc.h
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.cc
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
edia/base/fakemediaengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/basicpacketsocketfactory.cc
2p/base/basicpacketsocketfactory.h
2p/base/candidate.h
2p/base/common.h
2p/base/constants.cc
2p/base/constants.h
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransport.cc
2p/base/p2ptransport.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/packetsocketfactory.h
2p/base/parsing.cc
2p/base/parsing.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portallocator.cc
2p/base/portallocator.h
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy.h
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/pseudotcp.cc
2p/base/pseudotcp.h
2p/base/pseudotcp_unittest.cc
2p/base/rawtransport.cc
2p/base/rawtransport.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/relayserver_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/sessionclient.h
2p/base/sessiondescription.cc
2p/base/sessiondescription.h
2p/base/sessionid.h
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/stun.cc
2p/base/stun.h
2p/base/stun_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunrequest.cc
2p/base/stunrequest.h
2p/base/stunrequest_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/stunserver_unittest.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/testrelayserver.h
2p/base/teststunserver.h
2p/base/testturnserver.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.cc
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
2p/base/transportdescriptionfactory_unittest.cc
2p/base/transportinfo.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/base/udpport.h
2p/client/autoportallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
2p/client/httpportallocator.cc
2p/client/httpportallocator.h
2p/client/portallocator_unittest.cc
2p/client/sessionmanagertask.h
2p/client/sessionsendtask.h
2p/client/socketmonitor.cc
2p/client/socketmonitor.h
ession/media/audiomonitor.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediarecorder_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/media/rtcpmuxfilter.h
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
ession/media/typingmonitor_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
mpp/chatroommodule.h
mpp/chatroommoduleimpl.cc
mpp/constants.cc
mpp/constants.h
mpp/discoitemsquerytask.cc
mpp/discoitemsquerytask.h
mpp/fakexmppclient.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/hangoutpubsubclient_unittest.cc
mpp/iqtask.cc
mpp/iqtask.h
mpp/jid.cc
mpp/jid_unittest.cc
mpp/jingleinfotask.cc
mpp/jingleinfotask.h
mpp/module.h
mpp/moduleimpl.cc
mpp/moduleimpl.h
mpp/mucroomconfigtask.cc
mpp/mucroomconfigtask.h
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask.cc
mpp/mucroomdiscoverytask.h
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.cc
mpp/mucroomlookuptask.h
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask.cc
mpp/mucroomuniquehangoutidtask.h
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask.cc
mpp/pingtask.h
mpp/pingtask_unittest.cc
mpp/plainsaslhandler.h
mpp/presenceouttask.cc
mpp/presenceouttask.h
mpp/presencereceivetask.cc
mpp/presencereceivetask.h
mpp/presencestatus.cc
mpp/presencestatus.h
mpp/prexmppauth.h
mpp/pubsub_task.cc
mpp/pubsub_task.h
mpp/pubsubclient.cc
mpp/pubsubclient.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks.cc
mpp/pubsubtasks.h
mpp/pubsubtasks_unittest.cc
mpp/receivetask.cc
mpp/receivetask.h
mpp/rostermodule.h
mpp/rostermodule_unittest.cc
mpp/rostermoduleimpl.cc
mpp/rostermoduleimpl.h
mpp/saslcookiemechanism.h
mpp/saslmechanism.cc
mpp/saslplainmechanism.h
mpp/util_unittest.cc
mpp/util_unittest.h
mpp/xmppauth.cc
mpp/xmppauth.h
mpp/xmppclient.cc
mpp/xmppclient.h
mpp/xmppclientsettings.h
mpp/xmppengine.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmppengineimpl.h
mpp/xmppengineimpl_iq.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask.h
mpp/xmpplogintask_unittest.cc
mpp/xmpppump.cc
mpp/xmpppump.h
mpp/xmppsocket.h
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser_unittest.cc
mpp/xmpptask.cc
mpp/xmpptask.h
mpp/xmppthread.cc
mpp/xmppthread.h
|
| 81ddc78536585cb960699ed6e3c1a698645deb1e |
15-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77701902-> 77709729
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/videosource.cc
pp/webrtc/videosource_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
ession/media/channel_unittest.cc
|
| 1ecbe45c7e4c9142896cb2810d699558518f4f28 |
14-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77689511-> 77696841
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
edia/base/fakemediaengine.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/mediaengine.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
ession/media/call.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 43336b6b9f423778a2a97817e6c80ee2831322a8 |
14-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove unused (no-op) VideoOptions.
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
|
| a4351a045debf9f0450cf6cc8e1671094b21d19c |
14-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
libjingle: use _stricmp instead of deprecated stricmp.
BUG=N/A
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7447 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/transportdescription.cc
|
| 7fe1e03dd6da66401010119734245f114bf06645 |
14-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up external encoders.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/constants.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| f68cc0b0c3ce88c434afb8f55787b67b76c66ec6 |
14-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77554188-> 77629208
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7439 4adac7df-926f-26a2-2b94-8c16560cd09d
mllite/qname.cc
mllite/qname.h
mllite/qname_unittest.cc
mllite/xmlbuilder.cc
mllite/xmlbuilder.h
mllite/xmlbuilder_unittest.cc
mllite/xmlconstants.cc
mllite/xmlconstants.h
mllite/xmlelement.cc
mllite/xmlelement.h
mllite/xmlelement_unittest.cc
mllite/xmlnsstack.cc
mllite/xmlnsstack.h
mllite/xmlnsstack_unittest.cc
mllite/xmlparser.cc
mllite/xmlparser.h
mllite/xmlparser_unittest.cc
mllite/xmlprinter.cc
mllite/xmlprinter.h
mllite/xmlprinter_unittest.cc
|
| 1e6a5dd14e0c6d39995ecfdf14586f3b9503913e |
13-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Removes xmllite from talk/xmllite since webrtc/xmllite is used instead.
BUG=3379
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/23039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7436 4adac7df-926f-26a2-2b94-8c16560cd09d
mllite/qname.cc
mllite/qname.h
mllite/qname_unittest.cc
mllite/xmlbuilder.cc
mllite/xmlbuilder.h
mllite/xmlbuilder_unittest.cc
mllite/xmlconstants.cc
mllite/xmlconstants.h
mllite/xmlelement.cc
mllite/xmlelement.h
mllite/xmlelement_unittest.cc
mllite/xmlnsstack.cc
mllite/xmlnsstack.h
mllite/xmlnsstack_unittest.cc
mllite/xmlparser.cc
mllite/xmlparser.h
mllite/xmlparser_unittest.cc
mllite/xmlprinter.cc
mllite/xmlprinter.h
mllite/xmlprinter_unittest.cc
|
| 3c16d8bd1c0a3eea94a6678497eae0cf8e7f0187 |
13-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77414393-> 77554188
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoencoderfactory.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 3cefbc99f4cc2db744cb130ca629768401a59eb4 |
10-Oct-2014 |
xians@webrtc.org <xians@webrtc.org> |
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvoiceengine.h
|
| dae40dcde9f4ab4cb54a5a5f232fb225b625740f |
09-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Change setting VP8 codec specific info values by HW VP8 encoder
to follow SW implementation.
This fixes video freezing observed when connecting Android AppRtcDemo
on devices with hW encoder support with Chrome apprtc.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7414 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 95bacfed08a2e58e865bf6824a7428d8796aeb76 |
09-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Remove bad waiting code from video decoder release function.
Instead keep surface texture object alive while video codec
is re-initialized with a different resolution.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7401 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 97abeee2825ac93b62397feea74d0ad02d42540d |
09-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77263371-> 77296420
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediachannel.h
edia/base/rtpdataengine.h
edia/base/videoengine_unittest.h
edia/other/linphonemediaengine.h
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| 575d126a3d4a4bf6d43ea07189ac201f6bfe0798 |
08-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Protect send_/recv_streams_ in WebRtcVideoEngine2.
Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/22959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 742922b313baaebfbacf735287f9729a8bc6f8e0 |
07-Oct-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Make the media content send only if offerToReceive is false while local streams exist.
We previously do not add the media content if offerToReceive is false.
BUG=3833
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient_unittest.cc
|
| d6bda0950316f5c335f4471349bd518b2e7ba47f |
07-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Initialize sctp_paddrparams in OpenSctpSocket().
Addresses 'use-of-uninitialized-value' detected with MemorySanitizer.
params.spp_address.sa_family was used without being initialized before
when calling usrsctp_setsockopt with SCTP_PEER_ADDR_PARAMS.
R=jiayl@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7389 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine.cc
|
| 46ffc7087860789ddd2a794bfc1949e26ed3152b |
07-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7387 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 963b979510f6521fd69576f146235c6a5c0f8264 |
07-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove potential deadlock in WebRtcVideoEngine2.
Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.
R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999
Review URL: https://webrtc-codereview.appspot.com/26729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 6ed1cf49f0f80ee75f4615e301face39d328dd59 |
07-Oct-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Isolate: Remove use of --ignore_broken_items
BUG=chromium:395700
R=jam@chromium.org
Review URL: https://webrtc-codereview.appspot.com/30659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
|
| 528fc650d8546defdc0549a13b9c177d4981d7d1 |
06-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Fixing build issue with L-sdk
Based on https://codereview.appspot.com/153000043/
BUG=https://code.google.com/p/chromium/issues/detail?id=420293
R=niklas.enbom@webrtc.org, serya@chromium.org, yfriedman@chromium.org
Review URL: https://webrtc-codereview.appspot.com/29659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7374 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 42684be21b255e2b07eb154e6a2807fa2226167e |
03-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up CPU adaptation in WebRtcVideoEngine2.
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.
BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 25cc745d6b92962cec76abf30488e0a4cac36c98 |
02-Oct-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Switch to SW video decoder on Android after getting 2 or more
critical errors from HW decoder.
BUG=410730
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7368 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
|
| 4530b2ca48d914878a734f0663e8d566fda55c09 |
01-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7355 "Fix parallelization in libjingle_p2p_unittest."
Breaks waterfall.
TBR=pbos@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/22909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7357 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/relayserver_unittest.cc
2p/base/session_unittest.cc
|
| fd29205e6ee97e5ce96b833d57f32ca78b51dafc |
01-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix parallelization in libjingle_p2p_unittest.
Adding VirtualSocketServers to SessionTest and RelayServerTest to avoid
contention on real ports.
R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 out/Debug/libjingle_p2p_unittest
Review URL: https://webrtc-codereview.appspot.com/26679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7355 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/relayserver_unittest.cc
2p/base/session_unittest.cc
|
| 4cebd84c792309c98aed9ba05524ce051341268b |
01-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland "Remove DTMF status methods from Voice Engine" r7276
This reverts r7277.
TBR=henrika@webrtc.org,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 7aad5e5cced724c08b0af4bf85db446c5965ac76 |
30-Sep-2014 |
xians@webrtc.org <xians@webrtc.org> |
Revert 7338 "Fixed the android build by making the interface pur..."
> Fixed the android build by making the interface pure virtual.
>
> TBR=asapersson@webrtc.org, bjornv@webrtc.org,
>
> Review URL: https://webrtc-codereview.appspot.com/24789004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 90d1979d77ab07f9524e6e7738f135636c45bb74 |
30-Sep-2014 |
xians@webrtc.org <xians@webrtc.org> |
Fixed the android build by making the interface pure virtual.
TBR=asapersson@webrtc.org, bjornv@webrtc.org,
Review URL: https://webrtc-codereview.appspot.com/24789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 1795c358fcbad63ca0dea746dd74d073ae1faa22 |
30-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Add default implementation of Add/RemoveObserver.
Needed to remove Add/RemoveObserver from RTCVideoEncoderFactory in
Chromium before removing these completely. This is done to keep the
chromium.webrtc.fyi bots happy and to make rolling webrtc revisions
easier.
R=stefan@webrtc.org
BUG=3876
Review URL: https://webrtc-codereview.appspot.com/23839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7332 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoencoderfactory.h
|
| 8cad9432d554f9117faf147dcbaeabd6f7968e63 |
30-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Revert 7327 "Update isolate.gypi files + link to isolate_driver.py"
Breaks debug compilation (didn't run all trybots when testing this).
> Update isolate.gypi files + link to isolate_driver.py
>
> This updates the isolate.gypi copies we're forced to
> maintain in our code repo to Chromium revision c264a05.
>
> Since isolated testing is now using a new launch script
> in tools: isolate_driver.py, that is added to our links
> script.
>
> BUG=395700
> TESTED=Ran one of our tests with:
> ninja -C out/Release tools_unittests_run
> tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
>
> R=henrika@webrtc.org, jam@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/26649004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7328 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
|
| 02cd3067d2338239052d98a193b4fe13be5bbdfd |
30-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Update isolate.gypi files + link to isolate_driver.py
This updates the isolate.gypi copies we're forced to
maintain in our code repo to Chromium revision c264a05.
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that is added to our links
script.
BUG=395700
TESTED=Ran one of our tests with:
ninja -C out/Release tools_unittests_run
tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
R=henrika@webrtc.org, jam@chromium.org
Review URL: https://webrtc-codereview.appspot.com/26649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7327 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
|
| 359d72000483f4f44da864549eeebcfea70361f0 |
30-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Allow Android apps to set video renderer scaling type.
Also add type check for EGL context object received from apps and
switch to byte buffer video decoding if EGL context is incorrect
BUG=3851
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7326 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 7dfb7fa189701a57ad6399ad1472d6edd28b087c |
30-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Reland disallowing blocking calls on the worker thread.
This fixed the issue that invoking the call when the thread is not started.
BUG=3559
R=juberti@webrtc.org, thorcarpenter@google.com
Review URL: https://webrtc-codereview.appspot.com/24769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channelmanager.cc
|
| 626624061e0de73b346027660383d7ec006ae3b8 |
29-Sep-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Disable flaky tests:
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined
BUG=3871
R=henrike@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7323 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 34f2a9ea7245bac103fececfa53e92359680467a |
28-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Initialize SSL in unittest_main.cc.
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/rtpdataengine_unittest.cc
edia/sctp/sctpdataengine_unittest.cc
2p/base/dtlstransportchannel_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/relayport_unittest.cc
2p/base/relayserver_unittest.cc
2p/base/stunport_unittest.cc
2p/base/stunrequest_unittest.cc
2p/base/transportdescriptionfactory_unittest.cc
2p/base/turnport_unittest.cc
2p/client/portallocator_unittest.cc
ession/media/channel_unittest.cc
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient_unittest.cc
|
| bebc75e8bdd4172fec69ee376634ecbeb1191992 |
27-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix the duplicated candidate problem when using multiple STUN servers.
BUG=3723
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7312 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/teststunserver.h
2p/client/portallocator_unittest.cc
|
| a21d07160700d50514f16683a45d138d005c67c8 |
26-Sep-2014 |
thorcarpenter@google.com <thorcarpenter@google.com> |
Reverting part of
https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80
because of a major regression hanging the executable on start.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7309 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channelmanager.cc
|
| 05305116d6d1fe441614b553f201ef5c118220f3 |
25-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Explicitly initialize SSL for tests.
Adding missing SSL initialization/cleanups in
TransportDescriptionFactoryTest and MediaSessionTest.
These being missing prevent these tests from being run individually
without other tests preceding them that initialize SSL.
BUG=3860
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7300 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/transportdescriptionfactory_unittest.cc
ession/media/mediasessionclient_unittest.cc
|
| 3987b6de506a7e72a5bdfdf8c8ad9964705c5a28 |
24-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix a problem in Thread::Send.
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.
BUG=3559
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
ession/media/channelmanager.cc
|
| d60d79a14594cbc8266e4a50391ddbe64ed491f0 |
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Thread annotation of rtc::CriticalSection.
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.
This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.
R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
ession/media/mediamonitor.h
|
| 38344ed2806c8fed60d67d280ca44c32e36707c0 |
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.h
|
| 8166faeff3a1549c29a205b3a4f840b9b544c973 |
24-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Change Android video renderer to maintain video aspect
ratio when displaying camera or decoded video frames.
-
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
|
| 90668b1633794c80f8aa6999a4f6d4c5276922b5 |
23-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Switch HW video decoder to output byte buffers if video
renderer EGL context is not provided by app.
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 1b7dcc1647034c4a10b09170292d86231155e576 |
23-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 76169599-> 76176062
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 2c1bcea1bc5001bc2dc7d4eead87749f18eaadad |
23-Sep-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Enable ipv6 by default for webrtc under a Finch experiment.
Reapply 23529005 after fixing the build break issue (Chromium:582133002)
Committed: https://code.google.com/p/webrtc/source/detail?r=7253
Review URL: https://webrtc-codereview.appspot.com/23529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 3987f10c1142ffa07d749ce7b055b8a68892c19d |
23-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert "Remove DTMF status methods from Voice Engine" r7276
This change caused some trouble.
TBR=henrika@webrtc.org,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| bf7b9e0081233661ac0fe9500c0aa5b2aea70376 |
23-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove DTMF status methods from Voice Engine
These methods are not used.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 0a2087a7110e2455ce68f2c85068df5ae447508f |
23-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Skeleton for registering external encoders/decoders.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 83f95ba9a645099df5e19a91030029181d766b40 |
22-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove engine-level SetOptions.
Already removed in WebRtcVideoEngine.
R=andresp@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 64a2f10f4b566a91b358e77c4ecdf09ebb33ac59 |
22-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove Get/SetNetEQPlayoutMode APIs
These are not used anymore.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 97ed39344aa67d14b6a1a3591c1dbe14cf24c1a6 |
19-Sep-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Reapply 23529005 after fixing the build break issue (Chromium:582133002)
Review URL: https://webrtc-codereview.appspot.com/23529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| ed5ca1f1223c55f0c2026e096e6476127231fb37 |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75925673-> 75926712
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7252 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| c98f217c653d96cacd5dfb25758ccc6daa0dbbab |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75924589-> 75925673
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 0c9fe72b2136b86eaa57fe4a73bc9cafefdf80ff |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75922684-> 75924589
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7250 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.h
|
| ebf275733942865cdf7d54a8da93d401f9ded5e7 |
19-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix HW video decoder crash on some Android KK devices.
Remove direct access to decoder Java output buffer memory
when HW decoder is configured to decode to surface.
-
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30459005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7249 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| c1eebfa107d6e7c4f7be6502ec36125553bf578c |
19-Sep-2014 |
thorcarpenter@google.com <thorcarpenter@google.com> |
Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc.
R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com
Review URL: https://webrtc-codereview.appspot.com/22699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/sctp/sctpdataengine_unittest.cc
|
| e65812427db6609f4327dabc5336a9d07d7a182d |
19-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD.
Symbol LogcatTraceContext not defined.
Submitting on behalf of serya@.
Dup of https://webrtc-codereview.appspot.com/29529004/
TEST=Build target libjingle_peerconnection_javalib with applied CL https://codereview.chromium.org/551793003/
BUG=https://crbug.com/383418
R=serya@chromium.org
Review URL: https://webrtc-codereview.appspot.com/28529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| bbe0a8517d7f9da7aa779bff77cdbb70df358437 |
19-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Config struct for VideoEncoder.
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 6e5c78422d3b594f9c8bb4cce3e31da454d69711 |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75875619-> 75878731
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7235 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
|
| b5a5c44ef722762d4825c9dcde06b0db2bbb79a9 |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75865376-> 75875619
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7234 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| d7acf11e8d9f969c805ff5808324b761ffa74471 |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75854833-> 75865376
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| ccb3e3f3db449a434e7461361618d3a94b265106 |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75854418-> 75854833
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7232 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| dcc1f0426b3893f9c35e9a2674058e6ebe6b40e8 |
19-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75852725-> 75853560
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7231 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
2p/client/connectivitychecker.cc
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
|
| 0b435ba597eeb608a69e33c889d15dce55e5d1ea |
19-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
A few fixes to avoid crash in HW codec on device orientation change.
- Fix video encoder Reset() function to avoid setting codec
resolution to zero.
- Follow SW codec implementation and do not crash when frame
with the resolution different from the encoder resolution arrives.
Instead wait for at least 3 frames with new resolution and
re-initialize the codec. HW codec reset may take much longer
than SW codec, so these 3 frames threshold avoids resetting
codec when outstanding camera frame captured from previous device
orientation arrives.
- Plus some minor changes to make encoder reset/release
implementation closer to decoder implementation.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7230 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 83af77bf3c0918bd1d0361778ce304c06df6fa36 |
18-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Revert maximum video codec resolution on Android back to 720p again.
Some low end Android devices still have problems with 1080p support.
BUG=3757
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7228 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
|
| 933d88af58b00517570ef78f38852bfd7fb1bb02 |
18-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75818332-> 75837294
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7227 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 42731bdded823f29ce0984fb98905f9422bffc65 |
18-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Avoid writing a double/float to a string to avoid a crash.
BUG=crbug/367223
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7225 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port.cc
|
| 6cd6ba8ae016200a7a13b43294b8faf5d1d4affd |
18-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Expose VP8/H264 defaults through video_encoder.h.
Reduces code duplication quite a bit, these identical defaults were set
in quite a few different places.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=3070
Review URL: https://webrtc-codereview.appspot.com/19299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| ab071daab89462db77158e637ba059dba8c9ece7 |
18-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Split video_render_module implementation into default and internal implementation.
Targets must now link with implementation of their choice instead of at "gyp"-time.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common
Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests
GN changes:
- Not many since there is almost no test definitions.
Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.
Re-enable android tests by reverting 7026 (some tests left disabled).
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 369a637ac8032b3e41fd2b4f2f6b2ef49a447f02 |
18-Sep-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Implemented Network::GetBestIP() selection logic as following.
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7200
Committed: https://code.google.com/p/webrtc/source/detail?r=7201
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
2p/client/connectivitychecker.cc
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
|
| 3b67f8e0cab70a0ac164c55cd1e1765e9dc6eab5 |
17-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Enable HW video decoding on Qualcomm devices.
Parallel decoding and encoding problem is fixed now
(b/16353967), so it is possible to start using Qualcomm
VP8 HW decoder. Bitrate overshoots should be fixed as well.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7215 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 4a5061fbff6f752c0e25b6f4ffbe025823ca50e9 |
17-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
talk/p2p/base: removed unused variable "port_"
BUG=N/A
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7212 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/session_unittest.cc
|
| a74eda1b6f25234198dc5ebf433dc78c718a77e0 |
17-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Split video_capture_module specific implementation (external vs internal capture)
into its own targets. Dependencies must link directly with the desired one.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default/external capture implementation:
- anything dependent on webrtc_test_common
- anything dependent on video_engine_core
Targets linking with internal capture implementation:
- vie_auto_test
- anything dependent on webrtc_test_renderer
GN changes:
- Not many since there is almost no test definitions.
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 85ef770d92bb8b632bcdf73847c42c1461bd8922 |
17-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Split video engine android initialization into each internal module initialization.
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.
BUG=3768,3770
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=henrike@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| ab990ae43a2b84b103cb3c50bc38502375c13e68 |
17-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.
BUG=3070
TBR=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 6a9b155798ebe0854f035de61bae79460060f3d3 |
17-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75683337-> 75695882
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7206 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/asyncstuntcpsocket.h
2p/base/basicpacketsocketfactory.h
2p/base/packetsocketfactory.h
2p/client/autoportallocator.h
2p/client/sessionmanagertask.h
|
| a59c501c9928f69195ceae4b37fb399426df73a9 |
17-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Java VideoRenderer class may be backed by two different native
classes depending on type of rendering.
Fix crash in AppRtcDemo by calling correct destructor on exit.
BUG=
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7202 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
|
| 40c2aa36f21d311bba54f4af37d677f96404749d |
16-Sep-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Implemented Network::GetBestIP() selection logic as following.
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7200
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
2p/client/connectivitychecker.cc
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
|
| f8bff762d17720eee9410326ec2aa051979e4339 |
16-Sep-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Implemented Network::GetBestIP() selection logic as following.
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
2p/client/connectivitychecker.cc
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
|
| cddd17c0f89cfaa9d2f21118ae90b45dae3b4aee |
16-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Recreate VideoStreams when setting resolution.
Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 88e85ad64da6a8d949d0efa1f5456956ac65f9b9 |
16-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Add pbos@webrtc.org (myself) to talk/media/webrtc/.
Allows easier reviews of webrtcvideoengine2.cc and landing the new video
API on shorter review cycles.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/OWNERS
|
| 80132e4d70f3751f137d2b71b56cec9e306698f3 |
16-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75610402-> 75610402
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
ound/alsasoundsystem.cc
ound/alsasoundsystem.h
ound/alsasymboltable.cc
ound/alsasymboltable.h
ound/automaticallychosensoundsystem.h
ound/automaticallychosensoundsystem_unittest.cc
ound/linuxsoundsystem.cc
ound/linuxsoundsystem.h
ound/nullsoundsystem.cc
ound/nullsoundsystem.h
ound/nullsoundsystemfactory.cc
ound/nullsoundsystemfactory.h
ound/platformsoundsystem.cc
ound/platformsoundsystem.h
ound/platformsoundsystemfactory.cc
ound/platformsoundsystemfactory.h
ound/pulseaudiosoundsystem.cc
ound/pulseaudiosoundsystem.h
ound/pulseaudiosymboltable.cc
ound/pulseaudiosymboltable.h
ound/sounddevicelocator.h
ound/soundinputstreaminterface.h
ound/soundoutputstreaminterface.h
ound/soundsystemfactory.h
ound/soundsysteminterface.cc
ound/soundsysteminterface.h
ound/soundsystemproxy.cc
ound/soundsystemproxy.h
|
| 595b23c66fd120f4f2160f8c282f69869601cf61 |
16-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."
Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.
> Enable ipv6 by default for webrtc under a Finch experiment.
>
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
>
> Review URL: https://webrtc-codereview.appspot.com/23529005
TBR=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 6ae5a6d7fefff6759d338b5a3e3613e050ebaa62 |
16-Sep-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Add a target for the approved subset of rtc_base.
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)
This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.
The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.
BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.h
|
| 996784548d5024d97f795e330298ed71c68629e8 |
15-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
HW video decoding optimization to better support HD resolution:
- Change hw video decoder wrapper to allow to feed multiple input
and query for an output every 10 ms.
- Add an option to decode video frame into an Android surface object. Create
shared with video renderer EGL context and external texture on
video decoder thread.
- Support external texture rendering in Android renderer.
- Support TextureVideoFrame in Java and use it to pass texture from video decoder
to renderer.
- Fix HW encoder and decoder detection code to avoid query codec capabilities
from sw codecs.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| cd309e316801f4cd3cebc2a3654606db7a94828c |
15-Sep-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Enable ipv6 by default for webrtc under a Finch experiment.
BUG=413437 (chromium)
https://code.google.com/p/chromium/issues/detail?id=413437
Review URL: https://webrtc-codereview.appspot.com/23529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7184 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 000d86792d6fb57948ce60b3a3e9c8f34768f46c |
15-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make BW checks > 0 in peerconnection_unittest.cc.
These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.
R=stefan@webrtc.org
TBR=jiayl@webrtc.org, solenberg@webrtc.org
BUG=3817,chromium:375154
Review URL: https://webrtc-codereview.appspot.com/29479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 7f826350e365e7237dd127c1ef0e92b1fd7d1b8a |
15-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Stop building talk/xmllite since it is no longer used.
BUG=3379
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| a42a3ade541af4bc49e4e43a78bd886e3b140948 |
13-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75390072-> 75428737
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/devices/macdevicemanager.cc
edia/devices/macdevicemanagermm.mm
|
| 7e31197cb23e32dacaa7c0dd479d0cf21a23cfb8 |
13-Sep-2014 |
fbarchard@google.com <fbarchard@google.com> |
Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..."
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
> Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
>
> Breaks other repos.
>
> TBR=fbarchard@google.com
> BUG=N/A
>
> Review URL: https://webrtc-codereview.appspot.com/23639004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7173 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
|
| 192a54ff2f7e91e8ac3e21eff62fe8f5a8a21410 |
12-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Temporary revert maximum video codec resolution back to 1080p.
BUG=3757, 3738
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7171 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
|
| 3decd9b77609d2e6b89b08c0a80eeb06e4baaed2 |
12-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
Breaks other repos.
TBR=fbarchard@google.com
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/23639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7170 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
|
| ea77334c305a61e54ae5011d04e3a0364bfc6344 |
11-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75302540-> 75327856
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7160 4adac7df-926f-26a2-2b94-8c16560cd09d
mpp/asyncsocket.h
mpp/chatroommodule.h
mpp/jingleinfotask.h
mpp/module.h
mpp/moduleimpl.h
mpp/plainsaslhandler.h
mpp/presenceouttask.h
mpp/rostermodule.h
mpp/rostermoduleimpl.h
mpp/saslhandler.h
mpp/saslmechanism.h
mpp/xmppengine.h
mpp/xmppstanzaparser.h
|
| 1d8f780779f8426c60c55175a2ae7eaae83e7861 |
11-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Stop building talk/sound since it is no longer used.
BUG=N/A
R=pbos@webrtc.org
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7156 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| 1d53f64b0f42a32edff175004af3afd132bb1a8d |
11-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD.
webrtc::VideoEngine::SetAndroidObjects and webrtc::VoiceEngine::SetAndroidObjects
are not compatible with WEBRTC_CHROMIUM_BUILD. Since neither VoiceEngine nor VideoEngine
are needed at the time it's better to disable it completely.
BUG=https://crbug.com/412276
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7155 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 307d3dbdeed71d42edf38d3828081b11a5a416fb |
11-Sep-2014 |
henrikg@webrtc.org <henrikg@webrtc.org> |
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
Speculative revert, seems to be reason for flaky Win FYI bot compile break.
> Expose VideoEncoders with webrtc/video_encoder.h.
>
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
>
> BUG=3070
> R=mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21929004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| c665dcb2057e1ce9571963c449445cfaa8d6a3b0 |
11-Sep-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Revert 7145 "Stop building talk/sound since it is no longer used."
> Stop building talk/sound since it is no longer used.
>
> BUG=N/A
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/22319004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7148 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| 1972ff8a6e45f7ad3fb7e4ed51dc0135c72f6c9d |
11-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions). I've removed some of
these.
This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class. Removed "virtual" in those
cases.
BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcpassthroughrender.h
|
| 4c876453c8a8ef7aebc120566e28cb8c24eef91b |
11-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Stop building talk/sound since it is no longer used.
BUG=N/A
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7145 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| 3472dcd7b07f8e8fa4b0ac72a3821e4581e41ebb |
10-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Fix frame rate selection for Android camera.
- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.
BUG=2622
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/videosource.cc
|
| b2efb6771c3a5c492d8005a72b3c69cc18d1b408 |
10-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Put base tests in webrtc_tests.gyp
BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| b6d69282f563a3aa84a06ea20cfc133374c50a18 |
10-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Enable shared socket for TurnPort.
In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address.
TESTED=AppRtc loopback call generates both turn and stun candidates.
BUG=1746
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7138 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/portallocator.h
2p/base/stunport.h
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
|
| 5d639b3ef36c81a2330e5f0a4f7c119294400515 |
10-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75141932-> 75179475
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 7d4891d3f18861bdd5ec5d27409110cf3d110fa1 |
09-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7068
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/mediasession.cc
|
| 54cf1505e254f9e2b58b459ee4f32865696beacb |
09-Sep-2014 |
fbarchard@google.com <fbarchard@google.com> |
ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that.
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoframe.cc
|
| 22406fcc9bd70de7dcf2a536ed464d458d940b63 |
09-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7070
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
|
| 3d81b1b22a3ddc2047e95e74ca28dffa2bbfdaae |
09-Sep-2014 |
mallinath@webrtc.org <mallinath@webrtc.org> |
Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got
reverted due to some internal compile failures.
In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.
Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093
TBR=juberti@webrtc.org
BUG=1179
Review URL: https://webrtc-codereview.appspot.com/22329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
2p/base/portallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/portallocator_unittest.cc
|
| 4d19e05ab2d4f32484843d25fab809335b548230 |
09-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own.
This needs to happen sooner or later as if webrtc/base/checks.h happens to be included transitively here it would collide.
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7115 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| b420191743fc135222c862deeaa4cf9dec249fe3 |
09-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Expose VideoEncoders with webrtc/video_encoder.h.
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.
BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 8b0b21161abdcdc2f2528aadf25f1f8f5c99e8b2 |
09-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
TBR=mallinath@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/28419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
2p/base/portallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/portallocator_unittest.cc
|
| 7118e6166924b569805552253aebd8ae6e370bf3 |
08-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Finish work queue in SctpDataMediaChannelTest.
Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).
R=jiayl@webrtc.org
BUG=3608,chromium:375154
Review URL: https://webrtc-codereview.appspot.com/28399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine_unittest.cc
|
| 0e52772aa9d3dea65e2cd30187c4ff8e86f9eee4 |
08-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix a bot-breaking memory leak from early returning in ParseMediaDescription.
BUG=3791
R=henrike@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| c172320bd22311a0cf8c7c51c5c782e321622de1 |
08-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
This reverts commit r7068.
TBR=kjellander@webrtc.org
BUG=2108
Review URL: https://webrtc-codereview.appspot.com/23539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/mediasession.cc
|
| fd42f9dd6f56c2b9a615b92f5e85c0a6b0e47518 |
08-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74955991-> 75042522
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/mediaengine.h
edia/other/linphonemediaengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 7256d31d2864379d3299362f64b7a23741e67adb |
07-Sep-2014 |
mallinath@webrtc.org <mallinath@webrtc.org> |
Implementing ICE Transports type handling in libjingle transport.
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
2p/base/portallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/portallocator_unittest.cc
|
| cc060563f31e48980f9298abf9a0f90ba109d270 |
05-Sep-2014 |
thorcarpenter@google.com <thorcarpenter@google.com> |
Remove unnecessary include from testutils.cc.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/testutils.cc
|
| 992febb9978d2ded1a2c3c8a42ea18ee071ca3ae |
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74873066-> 74873164
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| a3344cfda4645b3a08fd58813a9ae7b33809c56b |
05-Sep-2014 |
thorcarpenter@google.com <thorcarpenter@google.com> |
Fix webrtcvideoframe tests.
R=fbarchard@google.com, harryjin@google.com, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/base/executablehelpers.h
edia/base/testutils.cc
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| ddb85ab85b233b4e038d7f0de093199199903a36 |
05-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added
BUG=3592
R=jiayl@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| af5fa952582f1cc12882d49cf3dbb4d5be2b3d2d |
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74857067-> 74860820
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 7e3bd3d7deaebf9b2fb51e4538a62d79ee0dec57 |
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74851128-> 74857067
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7083 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| bc6fa1876e2dd7f6bce845fc5d9f417f7a9b69c3 |
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74825992-> 74851128
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 818b7b3ac982e9e1f579904c5f160103da046dcf |
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74825084-> 74825992
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediaengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| dfbcf8161ef44bb952561a9a742c3d4e4487e405 |
05-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice.
BUG=3778
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7073 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling_unittest.cc
|
| f1427c673189971662d7cf2159195862640968f9 |
05-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7070 "TurnPort should retry allocation with a new address on error
STUN_ERROR_ALLOCATION_MISMATCH."
TBR=jiayl@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/15359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
|
| 4b234044d559e11afed48cf2e8335f4d44f4ba1c |
04-Sep-2014 |
glaznev@webrtc.org <glaznev@webrtc.org> |
Reduce maximum video resolution for Android.
HW video encoder and decoder can not be initialized
with 3840x2160 resolution.
BUG=3757,3738
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7071 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
|
| 574f2f60feaa41f4ca5d36381129066e6e8c25cb |
04-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
|
| 52055a276df3b0b0c3ed4c58ea74e0a4d8fe3891 |
04-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession_unittest.cc
ession/media/mediasession.cc
|
| ceb956b29dc28ffac03450240ce6a5741989a762 |
04-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Abort Negotiate() if DoCreateOffer() fails.
Addressing crash in test.
R=jiayl@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/19239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7066 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| bcb6bcfe6c50a1aa27fa243180887eaf9cf9a23b |
04-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove HybridVideoEngine.
This is currently unused dead code.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/hybridvideoengine_unittest.cc
|
| 95c245876616576dac1feb383ea02ae2880bab09 |
04-Sep-2014 |
thorcarpenter@google.com <thorcarpenter@google.com> |
* Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files.
"gcl try" fails to upload these large files so adding them independently.
R=andrew@webrtc.org, harryjin@google.com, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7050 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/testdata/faces.1280x720_P420.yuv
edia/testdata/faces_I400.jpg
edia/testdata/faces_I411.jpg
edia/testdata/faces_I420.jpg
edia/testdata/faces_I422.jpg
edia/testdata/faces_I444.jpg
|
| 609f987488fc15003b1603a10405c8696520c151 |
03-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74696326-> 74723281
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7047 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideoengine.cc
|
| fa4535b270d5f0d8575dffb4e60f1225751f77f0 |
03-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74694022-> 74696326
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
2p/base/constants.cc
2p/base/constants.h
2p/base/p2ptransport.cc
2p/base/parsing.h
2p/base/rawtransport.cc
2p/base/rawtransportchannel.cc
2p/base/session.h
2p/base/sessiondescription.cc
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/transport.cc
2p/base/transport_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient_unittest.cc
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
mpp/constants.cc
mpp/constants.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient_unittest.cc
mpp/jid.h
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask_unittest.cc
mpp/pubsub_task.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks_unittest.cc
mpp/rostermodule_unittest.cc
mpp/saslcookiemechanism.h
mpp/saslmechanism.cc
mpp/util_unittest.cc
mpp/xmppengine.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask_unittest.cc
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser.h
mpp/xmppstanzaparser_unittest.cc
|
| 26c0c41a06d77af54df547169d952a21319dea8c |
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Network up/down signaling in Call.
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| ebee401230bfb57a011e2e66a0f59d468c6941f0 |
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove flake in SendsLowerResolutionOnSmallerFrames.
Speculative fix for break on Linux64 Release. It looks like the second
frame is being dropped which is likely because the two frames are sent
too close to eachother. Adding a delay of 33ms in between them to make
sure the second one isn't dropped.
R=minyue@webrtc.org
TBR=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/22289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7043 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
|
| c4175b9fdf7d23eb58a044ff39e2b096f9091995 |
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set resolution based on incoming VideoFrames.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/17269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 72e448559ded94137473463e826a9f0df54caaeb |
03-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74628537-> 74648573
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7033 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/testutils.cc
|
| 90750482fadd3268c65534bcacab7c60ff6a1325 |
02-Sep-2014 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Remove deprecated RTCVideoRenderer constructor.
Removes -[RTCVideoRenderer initWithView]. Also, fix potential issue where we hold on to a video frame longer than the lifetime of its associated track.
BUG=3341
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7032 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCEAGLVideoView.m
pp/webrtc/objc/RTCNSGLVideoView.m
pp/webrtc/objc/RTCOpenGLVideoRenderer.mm
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/public/RTCVideoRenderer.h
xamples/objc/AppRTCDemo/ios/APPRTCViewController.m
|
| 9f341283f64d9b905c3883fd23988eb3d5fdcb8f |
02-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEngine::default_codec_format().
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/24399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7029 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.h
|
| 03655143dbb2fa9d6abacf386bdc29c37b211075 |
02-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove files from talk/PRESUBMIT.py.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7028 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| 44010f3e52b0907a73be0c1b41bf9c870446b8cb |
29-Aug-2014 |
thakis@chromium.org <thakis@chromium.org> |
win: Replace custom assert() macro with regular assert.h
The current code got added in libjingle r103; I don't see a good reason for it.
Things still build with plain old assert.h.
The custom assert was wrong: __debugbreak() is documented to return void,
so doing `cond ? true : __debugbreak()` shouldn't build (and it doesn't in
clang-cl). It's possible to make it build by writing
`cond ? true : (__debugbreak(), true)`, but just using the regular header
seems like a much better fix.
BUG=chromium:82385
Review URL: https://webrtc-codereview.appspot.com/19139004/
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7007 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/server/utils.h
|
| bc3f3339052d333cddb62ac984f964037569d430 |
29-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Add jiayl to talk OWNERS.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7006 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| e21cc9ae2a72b93863572d17bd5297aaa44923ea |
29-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated.
constraints . SetMandatoryReceiveAudio (false);
The problem is that webrtc::GetTrackIdBySsrc returns false if audio is not available. However it should continue and check for the video track.
BUG=webrtc:3755
R=jiayl@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7005 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
|
| 4431fd6ad59905ae3dfc9184762fe572f3c9bf97 |
28-Aug-2014 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org> |
Add 60 fps video support
R=henrike@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7000 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
|
| 1f8a23757af8ec10ba57fc14be221a5d53e8f2f1 |
28-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74235596-> 74297316
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6997 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
|
| 75c3ec17636319b77be24459efa529d14e76d7f1 |
27-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix data races during VideoAdapterTest tear-down.
Explicitly disconnect the VideoCapturer to avoid frames being
delivered during listener destruction. This manifested only on DrMemory
Full on Windows which I was able to repro locally.
BUG=3671
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6991 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoadapter_unittest.cc
|
| 573a1eef3dac80cfb5c65dfc7e6dbde9836b8d91 |
27-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74202294-> 74230205
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6990 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/devices/linuxdevicemanager.cc
edia/devices/linuxdevicemanager.h
|
| 00f11f5e2445d5ede48c394e8478308812bbbb71 |
27-Aug-2014 |
solenberg@webrtc.org <solenberg@webrtc.org> |
- Make local constant non-static.
- Remove spammy log line.
BUG=
R=henrike@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 7087857afd5d946877e8fbf086afa55880f24b6f |
26-Aug-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
implement handling ALTERNATE-SERVER response from turn protocol as
specified in RFC 5766, also created 2 test cases for both the normal
redirection case as well as when a pingpong situation happens, the
allocation should fail
BUG=1986 TURN ALTERNATE-SERVER support
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6985 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/stun.cc
2p/base/stun.h
2p/base/testturnserver.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
|
| 3533bfcb944d520b8491ff46e025c54334f15d66 |
26-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74132319-> 74133664
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6983 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 4470d78c9ba6a12ef25c6d66dcf4d3eab4e0e57c |
26-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74128148-> 74132319
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6982 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| f21ac1fd4658306026553d200d344b8553aec99b |
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix Win64 compile of videoadapter_unittest.cc.
Missed an typecast in videoadapter_unittest.cc in r6979 due to
tryservers being clogged and me waiting for a windows, linux, mac and
tsanv2 bot to finish was not enough. Committing fix straight away to
un-break tree.
TBR=tommi@webrtc.org
BUG=3671
Review URL: https://webrtc-codereview.appspot.com/18279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6980 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoadapter_unittest.cc
|
| c9b3f77e657517e30a3745e079ee64fe90409c5c |
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix data races in VideoAdapterTest.
Adressing clear races between the test thread and capturer thread shown
as heap-use-after-free in vpx_codec_destroy in
WebRtcVideoMediaChannelTest.SetSend (way later in the rest run).
When capturing a frame the test copied it to a separate frame that would
then be read by the test without synchronization, if the test didn't
manage to examine the frame in between captures the adapted frame would
be overwritten by the following frame during accesses to it.
The actual races are suppressed by race:webrtc/base/messagequeue.cc and
race:webrtc/base/thread.cc. These fixes reduce the suppression count
locally from around 3000 to 30 for VideoAdapterTest.*.
Also removing tsan suppressions for talk/base as it's been moved to
webrtc/base.
R=tommi@webrtc.org
BUG=3671
Review URL: https://webrtc-codereview.appspot.com/22169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6979 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoadapter_unittest.cc
|
| b648b9d85c5d07b0866ef45f5be587f71b0849b4 |
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove test constructor in WebRtcVideoEngine2.
Removes the need for ::Construct().
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| b96ea2aab5a5b2f170f374427f22159048bd1c1e |
26-Aug-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Remove former team members from OWNERS and WATCHLISTS
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@
BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
edia/webrtc/OWNERS
|
| 204cd560074cd6be857c32e0f8d69e77da810e57 |
25-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74064646-> 74072040
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6972 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| e9bfed0648656a22b41a9357e50a57d3c2d17e14 |
25-Aug-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Move constant so it is not stripped out for TSAN bots.
BUG=
R=henrike@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 857130fd5b9a1e828065ea914a00e7821cfade43 |
25-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74039473-> 74044292
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6970 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
|
| 6556a59db1efca3a4796d17de72420b0ddbcb29e |
25-Aug-2014 |
solenberg@webrtc.org <solenberg@webrtc.org> |
As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests.
Also, caused some issues with other peerconnection_unittest tests, so changed the design of those.
BUG=
R=kjellander@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6968 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| b4c7b09c1352174ecc1faf8c0cd93c66028a0485 |
25-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 3740d741068698baf987b1ced5ea485378e16d04 |
23-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73927658-> 73927775
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/call/callclient.cc
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybridvideoengine.h
edia/base/mediaengine.h
edia/other/linphonemediaengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
ession/media/call.cc
ession/media/call.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 309a611670cd71cb36d0b96a54e2db4fe65c22df |
23-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73891518-> 73927658
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6957 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 2b0554f0e744702a53936e69ee138002021f1e96 |
22-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73794259-> 73891518
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| 97fdeb8329cf5c328fa531c0a61c3dd181eb4833 |
22-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove static initializer in WebRtcVideoEngine2.
Blocks import into chromium.
R=tommi@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/18249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6954 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 7bd5fefb1734d7524d9bb36f0d4afa4afbfa16b8 |
21-Aug-2014 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Making sure muc members get recorded.
This is an upstream of a change I made; will describe in a separate
email thread.
Essentially, the members map wasn't getting populated in the callclient
example, so it was always empty. Now it will be populated correctly.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6950 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/call/callclient.cc
|
| 6908b84179e9302b5f8d8d5613af05a81d4fd184 |
20-Aug-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable two tests in TurnPortTest
The tests are flaky.
BUG=3720
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6934 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/turnport_unittest.cc
|
| 95bbd18696a27d87675675026f943f8f567c285f |
20-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73627179-> 73695227
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6933 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
|
| 5a60aed80f2b90f1b6c7e8af37e4c5bcc4ea02d1 |
19-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73626701-> 73627179
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6930 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
|
| 84532e59dd28f91e948d1f90e52acb23d3b26762 |
19-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73626167-> 73626701
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6929 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
|
| 0481f15f027fe1ef1768e90cc29362495114fb16 |
19-Aug-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73399579-> 73626167
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/webrtcsession.cc
ibjingle.gyp
edia/base/mediachannel.h
edia/base/mediaengine.cc
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| d5b292e450d64e42973815e25b4f7aa14a36dd28 |
19-Aug-2014 |
houssainy@google.com <houssainy@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Active connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc]
is now printed in the head-up display in Android appRTC.
This printing will be usefull in debugging switching ICE candidates.
R=andresp@webrtc.org, glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13189005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6927 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 353cd37ae9660ec9088e810d5fe68c92a8928266 |
15-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73370064-> 73399579
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6911 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/dtlstransportchannel.cc
|
| 5b06b06cc0ef5a051fa5b1ed687218a21639d93e |
15-Aug-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."
The bot that had the problem was using an old version of STL, so relanding.
> Revert 6863 "Refactor StatsCollector and associated types."
>
> Breaks chrome compilation on Mac:
>
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
> error: no matching constructor for initialization of
> 'webrtc::StatsReport'
> _Tp __x_copy = __x;
> ^ ~~~
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::_M_insert_aux' requested here
> _M_insert_aux(end(), __x);
> ^
> ../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::push_back' requested here
> reports.push_back(report1);
> ^
> ../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
> note: candidate constructor not viable: requires 0 arguments, but 1
> was provided
> StatsReport() : timestamp(0) {}
>
>
>
> > Refactor StatsCollector and associated types.
> > * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> > * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> > * Report ids are now const.
> > * Copying of data has been greatly reduced.
> > * This change includes preparation work for making GetStats fully async.
> >
> > This is a reland of r6778 which was reverted due to fyi bots failing.
> > I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> >
> > R=xians@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/15119004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21169004
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6908 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| c3df61e3510e67aae266e4196e3f98e48f4e83eb |
14-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73256845-> 73260148
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6898 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer.cc
|
| 22fa032f223e6b6210d569c8ae813c1a1a6edc07 |
14-Aug-2014 |
niklas.enbom@webrtc.org <niklas.enbom@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6863 "Refactor StatsCollector and associated types."
Breaks chrome compilation on Mac:
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
_Tp __x_copy = __x;
^ ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
_M_insert_aux(end(), __x);
^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
reports.push_back(report1);
^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
StatsReport() : timestamp(0) {}
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15119004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 449ad98aeb042ab09ae2da93ee94b9a90a1fc03c |
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73248599-> 73249894
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6896 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| ef8bb8d9b0bca0b1fd1ddb0a17df665e9dfaf9ad |
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure that muting muted streams succeeds.
We don't want to report an error here, and PeerConnection relies on
being able to mute already-muted streams (I hit an assert when testing
manually).
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6895 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| 432893a1002636aa83f7d356ed8e6f80f908d134 |
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove TODO saying to remove WebRtcVideoFrame.
Comment was added prematurely, there's no decision to get rid of this
type.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6894 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| b15dddf7ae34b3b5e5f268856e582264fce56011 |
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove files from talk/PRESUBMIT.py blacklist.
Many files can now be submitted here and do not have to be rolled in.
BUG=
R=henrike@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6893 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| d968dd039a4aa74afcaf50bf3be1be8e9707e548 |
13-Aug-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes failure triggered by include order re-ordering.
BUG=N/A
TBR=marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6892 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/audiotrackrenderer.h
pp/webrtc/datachannel.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource.h
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediastream_unittest.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/notifier.h
pp/webrtc/objc/RTCI420Frame.mm
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/RTCVideoSource.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/proxy_unittest.cc
pp/webrtc/remotevideocapturer.cc
pp/webrtc/remotevideocapturer_unittest.cc
pp/webrtc/sctputils_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/videosource.h
pp/webrtc/videosource_unittest.cc
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
xamples/call/call_main.cc
xamples/call/call_unittest.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/console.cc
xamples/call/console.h
xamples/call/friendinvitesendtask.cc
xamples/call/mediaenginefactory.cc
xamples/call/mucinviterecvtask.cc
xamples/call/mucinviterecvtask.h
xamples/call/mucinvitesendtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
xamples/objc/AppRTCDemo/ios/AppRTCDemo-Prefix.pch
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
xamples/peerconnection/client/main_wnd.cc
xamples/peerconnection/client/main_wnd.h
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/client/peer_connection_client.h
xamples/peerconnection/server/main.cc
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
edia/base/capturemanager.cc
edia/base/capturemanager.h
edia/base/capturemanager_unittest.cc
edia/base/capturerenderadapter.cc
edia/base/capturerenderadapter.h
edia/base/codec_unittest.cc
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/fakevideocapturer.h
edia/base/fakevideorenderer.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybriddataengine.h
edia/base/hybridvideoengine.h
edia/base/hybridvideoengine_unittest.cc
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/rtpdump.cc
edia/base/rtpdump_unittest.cc
edia/base/rtputils_unittest.cc
edia/base/streamparams.h
edia/base/streamparams_unittest.cc
edia/base/testutils.cc
edia/base/testutils.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videocommon_unittest.cc
edia/base/videoengine_unittest.h
edia/base/videoframe.cc
edia/base/videoframe_unittest.h
edia/base/videoprocessor.h
edia/base/voiceprocessor.h
edia/devices/carbonvideorenderer.cc
edia/devices/carbonvideorenderer.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/dummydevicemanager_unittest.cc
edia/devices/filevideocapturer.h
edia/devices/filevideocapturer_unittest.cc
edia/devices/gdivideorenderer.cc
edia/devices/gdivideorenderer.h
edia/devices/gtkvideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/devices/linuxdeviceinfo.cc
edia/devices/linuxdevicemanager.cc
edia/devices/linuxdevicemanager.h
edia/devices/macdevicemanager.cc
edia/devices/macdevicemanager.h
edia/devices/v4llookup.cc
edia/devices/win32devicemanager.h
edia/devices/yuvframescapturer.h
edia/other/linphonemediaengine.cc
edia/other/linphonemediaengine.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtcdeviceinfo.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcpassthroughrender_unittest.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtctexturevideoframe_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideocapturer_unittest.cc
edia/webrtc/webrtcvideoencoderfactory.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvie.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/basicpacketsocketfactory.cc
2p/base/candidate.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransport.cc
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/parsing.cc
2p/base/parsing.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portallocator.h
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.h
2p/base/pseudotcp_unittest.cc
2p/base/rawtransport.cc
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport_unittest.cc
2p/base/relayserver.h
2p/base/relayserver_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/sessiondescription.h
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/stun_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunrequest.h
2p/base/stunrequest_unittest.cc
2p/base/stunserver.h
2p/base/stunserver_unittest.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/testrelayserver.h
2p/base/teststunserver.h
2p/base/testturnserver.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory_unittest.cc
2p/base/transportinfo.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/client/autoportallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
2p/client/portallocator_unittest.cc
2p/client/sessionsendtask.h
2p/client/socketmonitor.h
ession/media/audiomonitor.cc
ession/media/audiomonitor.h
ession/media/bundlefilter.cc
ession/media/bundlefilter.h
ession/media/bundlefilter_unittest.cc
ession/media/call.cc
ession/media/call.h
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager_unittest.cc
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor_unittest.cc
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediamonitor.cc
ession/media/mediamonitor.h
ession/media/mediarecorder.cc
ession/media/mediarecorder.h
ession/media/mediarecorder_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/media/planarfunctions_unittest.cc
ession/media/rtcpmuxfilter.h
ession/media/rtcpmuxfilter_unittest.cc
ession/media/soundclip.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
ession/media/typingmonitor.cc
ession/media/typingmonitor.h
ession/media/typingmonitor_unittest.cc
ession/media/yuvscaler_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/securetunnelsessionclient.h
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
ound/alsasoundsystem.cc
ound/alsasoundsystem.h
ound/automaticallychosensoundsystem.h
ound/automaticallychosensoundsystem_unittest.cc
ound/nullsoundsystem.cc
ound/pulseaudiosoundsystem.cc
ound/pulseaudiosoundsystem.h
ound/soundsystemproxy.h
mllite/qname_unittest.cc
mllite/xmlbuilder.cc
mllite/xmlbuilder.h
mllite/xmlbuilder_unittest.cc
mllite/xmlelement.cc
mllite/xmlelement.h
mllite/xmlelement_unittest.cc
mllite/xmlnsstack.cc
mllite/xmlnsstack.h
mllite/xmlnsstack_unittest.cc
mllite/xmlparser.cc
mllite/xmlparser_unittest.cc
mllite/xmlprinter_unittest.cc
mpp/asyncsocket.h
mpp/chatroommodule_unittest.cc
mpp/chatroommoduleimpl.cc
mpp/constants.cc
mpp/discoitemsquerytask.cc
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/hangoutpubsubclient_unittest.cc
mpp/iqtask.cc
mpp/iqtask.h
mpp/jid.cc
mpp/jid.h
mpp/jid_unittest.cc
mpp/jingleinfotask.cc
mpp/module.h
mpp/moduleimpl.cc
mpp/moduleimpl.h
mpp/mucroomconfigtask.cc
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.cc
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask.cc
mpp/pingtask.h
mpp/pingtask_unittest.cc
mpp/plainsaslhandler.h
mpp/presenceouttask.cc
mpp/presenceouttask.h
mpp/presencereceivetask.cc
mpp/presencestatus.h
mpp/prexmppauth.h
mpp/pubsub_task.cc
mpp/pubsubclient.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks.h
mpp/pubsubtasks_unittest.cc
mpp/receivetask.cc
mpp/rostermodule_unittest.cc
mpp/rostermoduleimpl.cc
mpp/saslcookiemechanism.h
mpp/saslmechanism.cc
mpp/saslplainmechanism.h
mpp/util_unittest.cc
mpp/util_unittest.h
mpp/xmppauth.h
mpp/xmppclient.cc
mpp/xmppclient.h
mpp/xmppclientsettings.h
mpp/xmppengine.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmppengineimpl_iq.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask.h
mpp/xmpplogintask_unittest.cc
mpp/xmpppump.h
mpp/xmppsocket.h
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser.h
mpp/xmppstanzaparser_unittest.cc
mpp/xmpptask.cc
mpp/xmpptask.h
mpp/xmppthread.h
|
| 2c0fb05f1683b7a721072bdd93501b8afe164b9a |
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73221069-> 73222930
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.h
|
| 67f849575cc367ffd6385e464d4df42a87941a56 |
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73215194-> 73221069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6888 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 4eeeefebb20554aeef31aa9fcf5ee5280a7cb535 |
13-Aug-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73072800 -> 73215194
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6887 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 38d88816e395dfc32b355769f67a6f39c18bd511 |
13-Aug-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the audio source failure due to unsupported constraints.
Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.
This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.
BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback
R=henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource_unittest.cc
|
| e999bd087bcc7307c9e9b253e78837486213d124 |
13-Aug-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removing ASSERT for tcp candidate for port 0 and 9, as Android clients
may not be called with set_allow_tcp_listen(false).
This CL will also sends tcp candidate in RFC 6544 format.
BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| afb554f404d68e6f3ca5395216f776169370713d |
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move default-recv-channels to a separate class.
BUG=1788,3099
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6879 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| c3d2bd28a3e8badc434a5081dd36f4ac41b4e3f2 |
12-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix GetStats() crash.
GetStats() can be called before codecs are set and the underlying
webrtc::VideoSendStream is created, leading to a null-pointer
dereference.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6876 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 8d57f08902ce073095f1de7fa8bbdd0a1e5eac25 |
12-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73072800-> 73072800
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6873 4adac7df-926f-26a2-2b94-8c16560cd09d
hird_party/libudev/libudev.h
|
| 6ac22e6b47f9a6ed70b0a376984b39b9a745dd94 |
11-Aug-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| 730bf30da75514f22fc1869a93e130e582a8e045 |
11-Aug-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 7ec3f9f838f83d17f7ac1a938f174152fc3767a7 |
09-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a bug in parsing IceCandidate with IPV6 address.
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.
BUG=3669
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 9eabe5e9124723066ba8892ea850849ed9435dc6 |
09-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72931377-> 72931377
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6858 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/asyncfile.cc
ase/asyncfile.h
ase/asynchttprequest.cc
ase/asynchttprequest.h
ase/asynchttprequest_unittest.cc
ase/asyncinvoker-inl.h
ase/asyncinvoker.cc
ase/asyncinvoker.h
ase/asyncpacketsocket.h
ase/asyncresolverinterface.h
ase/asyncsocket.cc
ase/asyncsocket.h
ase/asynctcpsocket.cc
ase/asynctcpsocket.h
ase/asynctcpsocket_unittest.cc
ase/asyncudpsocket.cc
ase/asyncudpsocket.h
ase/asyncudpsocket_unittest.cc
ase/atomicops.h
ase/atomicops_unittest.cc
ase/autodetectproxy.cc
ase/autodetectproxy.h
ase/autodetectproxy_unittest.cc
ase/bandwidthsmoother.cc
ase/bandwidthsmoother.h
ase/bandwidthsmoother_unittest.cc
ase/base64.cc
ase/base64.h
ase/base64_unittest.cc
ase/basicdefs.h
ase/basictypes.h
ase/basictypes_unittest.cc
ase/bind.h
ase/bind.h.pump
ase/bind_unittest.cc
ase/buffer.h
ase/buffer_unittest.cc
ase/bytebuffer.cc
ase/bytebuffer.h
ase/bytebuffer_unittest.cc
ase/byteorder.h
ase/byteorder_unittest.cc
ase/callback.h
ase/callback.h.pump
ase/callback_unittest.cc
ase/checks.cc
ase/checks.h
ase/common.cc
ase/common.h
ase/compile_assert.h
ase/constructormagic.h
ase/cpumonitor.cc
ase/cpumonitor.h
ase/cpumonitor_unittest.cc
ase/crc32.cc
ase/crc32.h
ase/crc32_unittest.cc
ase/criticalsection.h
ase/criticalsection_unittest.cc
ase/cryptstring.h
ase/dbus.cc
ase/dbus.h
ase/dbus_unittest.cc
ase/diskcache.cc
ase/diskcache.h
ase/diskcache_win32.cc
ase/diskcache_win32.h
ase/dscp.h
ase/event.cc
ase/event.h
ase/event_unittest.cc
ase/fakecpumonitor.h
ase/fakenetwork.h
ase/fakesslidentity.h
ase/faketaskrunner.h
ase/filelock.cc
ase/filelock.h
ase/filelock_unittest.cc
ase/fileutils.cc
ase/fileutils.h
ase/fileutils_mock.h
ase/fileutils_unittest.cc
ase/firewallsocketserver.cc
ase/firewallsocketserver.h
ase/flags.cc
ase/flags.h
ase/gunit.h
ase/gunit_prod.h
ase/helpers.cc
ase/helpers.h
ase/helpers_unittest.cc
ase/httpbase.cc
ase/httpbase.h
ase/httpbase_unittest.cc
ase/httpclient.cc
ase/httpclient.h
ase/httpcommon-inl.h
ase/httpcommon.cc
ase/httpcommon.h
ase/httpcommon_unittest.cc
ase/httprequest.cc
ase/httprequest.h
ase/httpserver.cc
ase/httpserver.h
ase/httpserver_unittest.cc
ase/ifaddrs-android.cc
ase/ifaddrs-android.h
ase/iosfilesystem.mm
ase/ipaddress.cc
ase/ipaddress.h
ase/ipaddress_unittest.cc
ase/json.cc
ase/json.h
ase/json_unittest.cc
ase/latebindingsymboltable.cc
ase/latebindingsymboltable.cc.def
ase/latebindingsymboltable.h
ase/latebindingsymboltable.h.def
ase/latebindingsymboltable_unittest.cc
ase/libdbusglibsymboltable.cc
ase/libdbusglibsymboltable.h
ase/linked_ptr.h
ase/linux.cc
ase/linux.h
ase/linux_unittest.cc
ase/linuxfdwalk.c
ase/linuxfdwalk.h
ase/linuxfdwalk_unittest.cc
ase/linuxwindowpicker.cc
ase/linuxwindowpicker.h
ase/linuxwindowpicker_unittest.cc
ase/logging.cc
ase/logging.h
ase/logging_unittest.cc
ase/macasyncsocket.cc
ase/macasyncsocket.h
ase/maccocoasocketserver.h
ase/maccocoasocketserver.mm
ase/maccocoasocketserver_unittest.mm
ase/maccocoathreadhelper.h
ase/maccocoathreadhelper.mm
ase/macconversion.cc
ase/macconversion.h
ase/macsocketserver.cc
ase/macsocketserver.h
ase/macsocketserver_unittest.cc
ase/macutils.cc
ase/macutils.h
ase/macutils_unittest.cc
ase/macwindowpicker.cc
ase/macwindowpicker.h
ase/macwindowpicker_unittest.cc
ase/mathutils.h
ase/md5.cc
ase/md5.h
ase/md5digest.h
ase/md5digest_unittest.cc
ase/messagedigest.cc
ase/messagedigest.h
ase/messagedigest_unittest.cc
ase/messagehandler.cc
ase/messagehandler.h
ase/messagequeue.cc
ase/messagequeue.h
ase/messagequeue_unittest.cc
ase/move.h
ase/multipart.cc
ase/multipart.h
ase/multipart_unittest.cc
ase/nat_unittest.cc
ase/natserver.cc
ase/natserver.h
ase/natserver_main.cc
ase/natsocketfactory.cc
ase/natsocketfactory.h
ase/nattypes.cc
ase/nattypes.h
ase/nethelpers.cc
ase/nethelpers.h
ase/network.cc
ase/network.h
ase/network_unittest.cc
ase/nssidentity.cc
ase/nssidentity.h
ase/nssstreamadapter.cc
ase/nssstreamadapter.h
ase/nullsocketserver.h
ase/nullsocketserver_unittest.cc
ase/openssl.h
ase/openssladapter.cc
ase/openssladapter.h
ase/openssldigest.cc
ase/openssldigest.h
ase/opensslidentity.cc
ase/opensslidentity.h
ase/opensslstreamadapter.cc
ase/opensslstreamadapter.h
ase/optionsfile.cc
ase/optionsfile.h
ase/optionsfile_unittest.cc
ase/pathutils.cc
ase/pathutils.h
ase/pathutils_unittest.cc
ase/physicalsocketserver.cc
ase/physicalsocketserver.h
ase/physicalsocketserver_unittest.cc
ase/posix.cc
ase/posix.h
ase/profiler.cc
ase/profiler.h
ase/profiler_unittest.cc
ase/proxy_unittest.cc
ase/proxydetect.cc
ase/proxydetect.h
ase/proxydetect_unittest.cc
ase/proxyinfo.cc
ase/proxyinfo.h
ase/proxyserver.cc
ase/proxyserver.h
ase/ratelimiter.cc
ase/ratelimiter.h
ase/ratelimiter_unittest.cc
ase/ratetracker.cc
ase/ratetracker.h
ase/ratetracker_unittest.cc
ase/refcount.h
ase/referencecountedsingletonfactory.h
ase/referencecountedsingletonfactory_unittest.cc
ase/rollingaccumulator.h
ase/rollingaccumulator_unittest.cc
ase/safe_conversions.h
ase/safe_conversions_impl.h
ase/schanneladapter.cc
ase/schanneladapter.h
ase/scoped_autorelease_pool.h
ase/scoped_autorelease_pool.mm
ase/scoped_ptr.h
ase/scoped_ref_ptr.h
ase/scopedptrcollection.h
ase/scopedptrcollection_unittest.cc
ase/sec_buffer.h
ase/sha1.cc
ase/sha1.h
ase/sha1digest.h
ase/sha1digest_unittest.cc
ase/sharedexclusivelock.cc
ase/sharedexclusivelock.h
ase/sharedexclusivelock_unittest.cc
ase/signalthread.cc
ase/signalthread.h
ase/signalthread_unittest.cc
ase/sigslot.h
ase/sigslot_unittest.cc
ase/sigslotrepeater.h
ase/sigslottester.h
ase/sigslottester.h.pump
ase/sigslottester_unittest.cc
ase/socket.h
ase/socket_unittest.cc
ase/socket_unittest.h
ase/socketadapters.cc
ase/socketadapters.h
ase/socketaddress.cc
ase/socketaddress.h
ase/socketaddress_unittest.cc
ase/socketaddresspair.cc
ase/socketaddresspair.h
ase/socketfactory.h
ase/socketpool.cc
ase/socketpool.h
ase/socketserver.h
ase/socketstream.cc
ase/socketstream.h
ase/ssladapter.cc
ase/ssladapter.h
ase/sslconfig.h
ase/sslfingerprint.cc
ase/sslfingerprint.h
ase/sslidentity.cc
ase/sslidentity.h
ase/sslidentity_unittest.cc
ase/sslroots.h
ase/sslsocketfactory.cc
ase/sslsocketfactory.h
ase/sslstreamadapter.cc
ase/sslstreamadapter.h
ase/sslstreamadapter_unittest.cc
ase/sslstreamadapterhelper.cc
ase/sslstreamadapterhelper.h
ase/stream.cc
ase/stream.h
ase/stream_unittest.cc
ase/stringdigest.h
ase/stringencode.cc
ase/stringencode.h
ase/stringencode_unittest.cc
ase/stringutils.cc
ase/stringutils.h
ase/stringutils_unittest.cc
ase/systeminfo.cc
ase/systeminfo.h
ase/systeminfo_unittest.cc
ase/task.cc
ase/task.h
ase/task_unittest.cc
ase/taskparent.cc
ase/taskparent.h
ase/taskrunner.cc
ase/taskrunner.h
ase/template_util.h
ase/testbase64.h
ase/testclient.cc
ase/testclient.h
ase/testclient_unittest.cc
ase/testechoserver.h
ase/testutils.h
ase/thread.cc
ase/thread.h
ase/thread_unittest.cc
ase/timeutils.cc
ase/timeutils.h
ase/timeutils_unittest.cc
ase/timing.cc
ase/timing.h
ase/transformadapter.cc
ase/transformadapter.h
ase/unittest_main.cc
ase/unixfilesystem.cc
ase/unixfilesystem.h
ase/urlencode.cc
ase/urlencode.h
ase/urlencode_unittest.cc
ase/versionparsing.cc
ase/versionparsing.h
ase/versionparsing_unittest.cc
ase/virtualsocket_unittest.cc
ase/virtualsocketserver.cc
ase/virtualsocketserver.h
ase/win32.cc
ase/win32.h
ase/win32_unittest.cc
ase/win32filesystem.cc
ase/win32filesystem.h
ase/win32regkey.cc
ase/win32regkey.h
ase/win32regkey_unittest.cc
ase/win32securityerrors.cc
ase/win32socketinit.cc
ase/win32socketinit.h
ase/win32socketserver.cc
ase/win32socketserver.h
ase/win32socketserver_unittest.cc
ase/win32toolhelp.h
ase/win32toolhelp_unittest.cc
ase/win32window.cc
ase/win32window.h
ase/win32window_unittest.cc
ase/win32windowpicker.cc
ase/win32windowpicker.h
ase/win32windowpicker_unittest.cc
ase/window.h
ase/windowpicker.h
ase/windowpicker_unittest.cc
ase/windowpickerfactory.h
ase/winfirewall.cc
ase/winfirewall.h
ase/winfirewall_unittest.cc
ase/winping.cc
ase/winping.h
ase/worker.cc
ase/worker.h
|
| 2d60c5e8bcac85e9388e093bae91ecc829eabcea |
09-Aug-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Encoding and Decoding of TCP candidates as defined in RFC 6544.
R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204
Review URL: https://webrtc-codereview.appspot.com/21479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
2p/base/candidate.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/relayport.cc
2p/base/stunport.cc
2p/base/tcpport.cc
2p/base/transport.cc
2p/base/turnport.cc
2p/client/connectivitychecker_unittest.cc
|
| 53df88c1bcd42d79d178f8c8da8d4d620f1c12cf |
08-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72847605-> 72850595
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| 65b98d12c3b6b9ca0ded669d0a0811d2bb1712b3 |
08-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72839629-> 72847605
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| c8554be6dd41c6fe48eb6b09e9cfc0fb0064a7ff |
07-Aug-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support for TURN/TLS.
Wrap the socket in an SSL adapter, then simply call StartSSL() on the
SSLAdapter instance.
Cloned from: https://webrtc-codereview.appspot.com/21799004/
R=juberti@chromium.org, juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/14059004
Patch from Manish Jethani <manish.jethani@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6852 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/basicpacketsocketfactory.cc
|
| cb46de24fb4259876fa2d87678f686bb3ae49e04 |
07-Aug-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add new OWNERS file to talk/examples.
R=juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/15039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6851 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/OWNERS
|
| 5b1ebacca2c29d73a5f3ab388b4b2a0a8e114c76 |
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72820109-> 72822008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| d509678a4e5ba4c3047d80744e103b675d8c7c88 |
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72819313-> 72820109
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 94b996cc181b02d986f002230497bb2b28762060 |
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72785516-> 72819313
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 476efa203160463dafc2d5bf9b8a675df44d2df5 |
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72785180-> 72785516
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 4f0d401faecf5d8a4c82e6e2223651ef13ad8e31 |
07-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72682155-> 72785180
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6841 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
edia/base/fakevideocapturer.h
edia/base/mutedvideocapturer.cc
edia/base/mutedvideocapturer.h
edia/base/mutedvideocapturer_unittest.cc
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoengine_unittest.h
edia/base/videoframefactory.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoframefactory.cc
edia/webrtc/webrtcvideoframefactory.h
|
| 56d8e05238a46bfc51fcb804bc1f5477dfefcc14 |
06-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
A followup to r6828 to fix a condition check in mediasession.cc.
BUG=2395
R=juberti@chromium.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6832 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/mediasession.cc
|
| 624a504f5ba0c2ca2a9a138e6d3ed1c1937b8df4 |
06-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72659510-> 72673987
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6829 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine.cc
|
| e7d47a1473e885a57986dcdbf06e7e1d25226ca6 |
05-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Maintain the order of the m-lines in CreateOffer and CreateAnswer.
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.
BUG=2395
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/webrtcsdp_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| 8e885990aed58a26ae1b27dd7547536393879a7c |
05-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72566057-> 72591796
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6824 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
edia/base/fakevideocapturer.h
edia/base/mutedvideocapturer.cc
edia/base/mutedvideocapturer.h
edia/base/mutedvideocapturer_unittest.cc
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoengine_unittest.h
edia/base/videoframefactory.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoframefactory.cc
edia/webrtc/webrtcvideoframefactory.h
|
| b18bf5e47d1db8ca563c9c6f12e77f9cd63879d4 |
04-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer.
Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.
BUG=3282
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
|
| a27342b7afb03906813d3efb214d8bae7ad0b7b8 |
04-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72446860-> 72550257
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6818 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| e0d03f13e4cfc5b822145597d40da9b8a8f95146 |
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72443101-> 72446860
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 6e203d50a3ecccc0524d36867761f80c12e0c56f |
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72442050-> 72443101
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 52148c2f74fe455ee126d24ec57a8bfc7cc87404 |
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72430895-> 72442050
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 7cb60ccae137d8db99e00ed2e073a00f110ccc57 |
02-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72407428-> 72430895
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/base/device.h
edia/base/fakescreencapturerfactory.h
edia/base/videocapturerfactory.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/webrtc/webrtcvideocapturerfactory.cc
edia/webrtc/webrtcvideocapturerfactory.h
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 3bc48247b7d009f708550cd5c0470038f9045d08 |
01-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72403605-> 72407428
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6811 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
edia/base/fakevideocapturer.h
edia/base/mutedvideocapturer.cc
edia/base/mutedvideocapturer.h
edia/base/mutedvideocapturer_unittest.cc
edia/base/videoadapter_unittest.cc
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoengine_unittest.h
edia/base/videoframefactory.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoframefactory.cc
edia/webrtc/webrtcvideoframefactory.h
|
| 6955213ecacdef941c259e8be0685b18e69b2252 |
01-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72389720-> 72403605
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6810 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 42d65ce8d75005758e69c5f6a84684d2b3132e53 |
01-Aug-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots.
TBR=hellner
BUG=
Review URL: https://webrtc-codereview.appspot.com/18969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6809 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/fakesslidentity.h
|
| 1a678c61f1bc001e50771a46c5714e56238a1c10 |
01-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72320533-> 72380285
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6808 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 6b21b710686b017badb7853acf5d20ca92e162cd |
31-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72205295-> 72320533
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcexport.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| d9843da9ee072ddb8d583de869a622151d914f54 |
30-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
libjingle: stop building files in talk/base as they are no longer used as of r6799
BUG=3379
R=thorcarpenter@google.com
Review URL: https://webrtc-codereview.appspot.com/16189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6802 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 48305f5f4c665d3e11d2e414570ffe8494f13709 |
30-Jul-2014 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013.
BUG=3584
TESTED=python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Release
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6801 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/audiotrack.cc
pp/webrtc/audiotrack.h
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/dtmfsender.cc
pp/webrtc/dtmfsender.h
pp/webrtc/dtmfsender_unittest.cc
pp/webrtc/dtmfsenderinterface.h
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/Logging.java
pp/webrtc/jsep.h
pp/webrtc/jsepicecandidate.cc
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/jsepsessiondescription.h
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource.h
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediastream.cc
pp/webrtc/mediastream.h
pp/webrtc/mediastream_unittest.cc
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamproxy.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/notifier.h
pp/webrtc/objc/RTCAudioTrack+Internal.h
pp/webrtc/objc/RTCAudioTrack.mm
pp/webrtc/objc/RTCDataChannel+Internal.h
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/objc/RTCI420Frame.mm
pp/webrtc/objc/RTCMediaConstraints.mm
pp/webrtc/objc/RTCMediaSource+Internal.h
pp/webrtc/objc/RTCMediaSource.mm
pp/webrtc/objc/RTCMediaStream+Internal.h
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCMediaStreamTrack+Internal.h
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCPeerConnection+Internal.h
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCVideoCapturer.mm
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/RTCVideoSource+Internal.h
pp/webrtc/objc/RTCVideoSource.mm
pp/webrtc/objc/RTCVideoTrack+Internal.h
pp/webrtc/objc/RTCVideoTrack.mm
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/objctests/mac/main.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/portallocatorfactory.cc
pp/webrtc/portallocatorfactory.h
pp/webrtc/proxy.h
pp/webrtc/proxy_unittest.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/remotevideocapturer.cc
pp/webrtc/remotevideocapturer_unittest.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/streamcollection.h
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/test/fakedtlsidentityservice.h
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
pp/webrtc/videosource.cc
pp/webrtc/videosource.h
pp/webrtc/videosource_unittest.cc
pp/webrtc/videotrack.cc
pp/webrtc/videotrack.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
uild/common.gypi
xamples/call/call_main.cc
xamples/call/call_unittest.cc
xamples/call/callclient.cc
xamples/call/callclient.h
xamples/call/callclient_unittest.cc
xamples/call/console.cc
xamples/call/console.h
xamples/call/mediaenginefactory.cc
xamples/call/mucinviterecvtask.h
xamples/call/presencepushtask.cc
xamples/call/presencepushtask.h
xamples/login/login_main.cc
xamples/peerconnection/client/conductor.cc
xamples/peerconnection/client/conductor.h
xamples/peerconnection/client/defaults.cc
xamples/peerconnection/client/defaults.h
xamples/peerconnection/client/flagdefs.h
xamples/peerconnection/client/linux/main.cc
xamples/peerconnection/client/linux/main_wnd.cc
xamples/peerconnection/client/linux/main_wnd.h
xamples/peerconnection/client/main.cc
xamples/peerconnection/client/main_wnd.cc
xamples/peerconnection/client/main_wnd.h
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/client/peer_connection_client.h
xamples/peerconnection/server/main.cc
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/capturemanager.cc
edia/base/capturemanager.h
edia/base/capturemanager_unittest.cc
edia/base/capturerenderadapter.cc
edia/base/capturerenderadapter.h
edia/base/codec.cc
edia/base/codec_unittest.cc
edia/base/cpuid.h
edia/base/cpuid_unittest.cc
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/fakevideocapturer.h
edia/base/fakevideorenderer.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybriddataengine.h
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/hybridvideoengine_unittest.cc
edia/base/mediachannel.h
edia/base/mediacommon.h
edia/base/mediaengine.h
edia/base/mutedvideocapturer.cc
edia/base/mutedvideocapturer.h
edia/base/mutedvideocapturer_unittest.cc
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/rtpdump.cc
edia/base/rtpdump.h
edia/base/rtpdump_unittest.cc
edia/base/rtputils.cc
edia/base/rtputils.h
edia/base/rtputils_unittest.cc
edia/base/screencastid.h
edia/base/streamparams.h
edia/base/streamparams_unittest.cc
edia/base/testutils.cc
edia/base/testutils.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videocommon.cc
edia/base/videocommon.h
edia/base/videocommon_unittest.cc
edia/base/videoengine_unittest.h
edia/base/videoframe.cc
edia/base/videoframe.h
edia/base/videoframe_unittest.h
edia/base/videoprocessor.h
edia/base/videorenderer.h
edia/base/voiceprocessor.h
edia/base/yuvframegenerator.cc
edia/base/yuvframegenerator.h
edia/devices/carbonvideorenderer.cc
edia/devices/carbonvideorenderer.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/devicemanager_unittest.cc
edia/devices/dummydevicemanager_unittest.cc
edia/devices/fakedevicemanager.h
edia/devices/filevideocapturer.cc
edia/devices/filevideocapturer.h
edia/devices/filevideocapturer_unittest.cc
edia/devices/gdivideorenderer.cc
edia/devices/gdivideorenderer.h
edia/devices/gtkvideorenderer.h
edia/devices/libudevsymboltable.cc
edia/devices/libudevsymboltable.h
edia/devices/linuxdeviceinfo.cc
edia/devices/linuxdevicemanager.cc
edia/devices/linuxdevicemanager.h
edia/devices/macdevicemanager.cc
edia/devices/macdevicemanager.h
edia/devices/macdevicemanagermm.mm
edia/devices/mobiledevicemanager.cc
edia/devices/v4llookup.cc
edia/devices/win32devicemanager.cc
edia/devices/win32devicemanager.h
edia/devices/yuvframescapturer.cc
edia/devices/yuvframescapturer.h
edia/other/linphonemediaengine.cc
edia/other/linphonemediaengine.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtccommon.h
edia/webrtc/fakewebrtcdeviceinfo.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcpassthroughrender.cc
edia/webrtc/webrtcpassthroughrender.h
edia/webrtc/webrtcpassthroughrender_unittest.cc
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtctexturevideoframe_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideocapturer_unittest.cc
edia/webrtc/webrtcvideodecoderfactory.h
edia/webrtc/webrtcvideoencoderfactory.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvie.h
edia/webrtc/webrtcvoe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/basicpacketsocketfactory.cc
2p/base/basicpacketsocketfactory.h
2p/base/candidate.h
2p/base/common.h
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransport.cc
2p/base/p2ptransport.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/packetsocketfactory.h
2p/base/parsing.cc
2p/base/parsing.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portallocator.h
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy.h
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/pseudotcp.cc
2p/base/pseudotcp.h
2p/base/pseudotcp_unittest.cc
2p/base/rawtransport.cc
2p/base/rawtransport.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/relayserver_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/sessiondescription.h
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
2p/base/sessionmessages.cc
2p/base/sessionmessages.h
2p/base/stun.cc
2p/base/stun.h
2p/base/stun_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunrequest.cc
2p/base/stunrequest.h
2p/base/stunrequest_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/stunserver_unittest.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/testrelayserver.h
2p/base/teststunserver.h
2p/base/testturnserver.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
2p/base/transportdescriptionfactory_unittest.cc
2p/base/transportinfo.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/client/autoportallocator.h
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/fakeportallocator.h
2p/client/httpportallocator.cc
2p/client/httpportallocator.h
2p/client/portallocator_unittest.cc
2p/client/sessionsendtask.h
2p/client/socketmonitor.cc
2p/client/socketmonitor.h
ession/media/audiomonitor.cc
ession/media/audiomonitor.h
ession/media/bundlefilter.cc
ession/media/bundlefilter.h
ession/media/bundlefilter_unittest.cc
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor.h
ession/media/currentspeakermonitor_unittest.cc
ession/media/externalhmac.cc
ession/media/externalhmac.h
ession/media/mediamessages.cc
ession/media/mediamessages.h
ession/media/mediamessages_unittest.cc
ession/media/mediamonitor.cc
ession/media/mediamonitor.h
ession/media/mediarecorder.cc
ession/media/mediarecorder.h
ession/media/mediarecorder_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/mediasessionclient_unittest.cc
ession/media/planarfunctions_unittest.cc
ession/media/rtcpmuxfilter.cc
ession/media/rtcpmuxfilter.h
ession/media/rtcpmuxfilter_unittest.cc
ession/media/soundclip.cc
ession/media/soundclip.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
ession/media/typewrapping.h.pump
ession/media/typingmonitor.cc
ession/media/typingmonitor.h
ession/media/typingmonitor_unittest.cc
ession/media/yuvscaler_unittest.cc
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
ession/tunnel/securetunnelsessionclient.cc
ession/tunnel/securetunnelsessionclient.h
ession/tunnel/tunnelsessionclient.cc
ession/tunnel/tunnelsessionclient.h
ession/tunnel/tunnelsessionclient_unittest.cc
ound/alsasoundsystem.cc
ound/alsasoundsystem.h
ound/alsasymboltable.cc
ound/alsasymboltable.h
ound/automaticallychosensoundsystem.h
ound/automaticallychosensoundsystem_unittest.cc
ound/nullsoundsystem.cc
ound/platformsoundsystem.cc
ound/pulseaudiosoundsystem.cc
ound/pulseaudiosoundsystem.h
ound/pulseaudiosymboltable.cc
ound/pulseaudiosymboltable.h
ound/sounddevicelocator.h
ound/soundinputstreaminterface.h
ound/soundoutputstreaminterface.h
ound/soundsystemfactory.h
ound/soundsysteminterface.h
ound/soundsystemproxy.h
mllite/qname_unittest.cc
mllite/xmlbuilder.cc
mllite/xmlbuilder.h
mllite/xmlbuilder_unittest.cc
mllite/xmlelement.cc
mllite/xmlelement.h
mllite/xmlelement_unittest.cc
mllite/xmlnsstack.h
mllite/xmlnsstack_unittest.cc
mllite/xmlparser.cc
mllite/xmlparser_unittest.cc
mllite/xmlprinter_unittest.cc
mpp/asyncsocket.h
mpp/chatroommodule_unittest.cc
mpp/chatroommoduleimpl.cc
mpp/constants.cc
mpp/discoitemsquerytask.cc
mpp/fakexmppclient.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/hangoutpubsubclient_unittest.cc
mpp/iqtask.h
mpp/jid.cc
mpp/jid.h
mpp/jid_unittest.cc
mpp/jingleinfotask.cc
mpp/jingleinfotask.h
mpp/moduleimpl.cc
mpp/mucroomconfigtask.cc
mpp/mucroomconfigtask_unittest.cc
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.cc
mpp/mucroomlookuptask_unittest.cc
mpp/mucroomuniquehangoutidtask_unittest.cc
mpp/pingtask.cc
mpp/pingtask.h
mpp/pingtask_unittest.cc
mpp/plainsaslhandler.h
mpp/presenceouttask.cc
mpp/presencereceivetask.cc
mpp/presencereceivetask.h
mpp/prexmppauth.h
mpp/pubsub_task.cc
mpp/pubsubclient.h
mpp/pubsubclient_unittest.cc
mpp/pubsubstateclient.h
mpp/pubsubtasks.h
mpp/pubsubtasks_unittest.cc
mpp/rostermodule_unittest.cc
mpp/rostermoduleimpl.cc
mpp/rostermoduleimpl.h
mpp/saslmechanism.cc
mpp/saslplainmechanism.h
mpp/util_unittest.cc
mpp/xmppauth.cc
mpp/xmppauth.h
mpp/xmppclient.cc
mpp/xmppclient.h
mpp/xmppclientsettings.h
mpp/xmppengine_unittest.cc
mpp/xmppengineimpl.cc
mpp/xmppengineimpl.h
mpp/xmppengineimpl_iq.cc
mpp/xmpplogintask.cc
mpp/xmpplogintask.h
mpp/xmpplogintask_unittest.cc
mpp/xmpppump.cc
mpp/xmpppump.h
mpp/xmppsocket.cc
mpp/xmppsocket.h
mpp/xmppstanzaparser.cc
mpp/xmppstanzaparser_unittest.cc
mpp/xmpptask.h
mpp/xmppthread.cc
mpp/xmppthread.h
|
| 51c5508bf1489f6b65bde2373b97cdf2e3af2426 |
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72016417-> 72097588
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
xamples/call/callclient.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| 8aed9458428e715372cbabbbd6ea376eeb805dd9 |
26-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove a disabled test.
ConstrainsSetCodecsAccordingToEncoderConfig has been removed from
webrtcvideoengine_unittest.cc, removing this one as well.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6789 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| af9e7943d1b3e38b1d92d77fd91eeaf50148c3b9 |
25-Jul-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix compilation on windows with clang, indentation cleanups
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6779
Review URL: https://webrtc-codereview.appspot.com/18849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6786 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/httpcommon.cc
ase/schanneladapter.cc
|
| 257e130a1639febeb3ffc4d42943be3cb58151c7 |
25-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set NACK/REMB when setting receive codecs.
Enabling an additional test to ensure that REMB can be both enabled and
disabled by setting VideoCodecs.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6785 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 185636cf708890c04d540be1cf4be4a867c984c0 |
25-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert of 6778 "Refactor StatsCollector and associated types."
Breakes FYI bots.
BUG=N/A
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6783 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 2386882266c7a6a23aa11df0de6f52719eb0e7c3 |
25-Jul-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Fix compilation on windows with clang, indentation cleanups"
This reverts commit f628eaedfeea97e13c63c78dd42f2b1c76723619.
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/13069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6780 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/httpcommon.cc
ase/schanneladapter.cc
|
| a44fce59200b7cfdf3e38a6a97598d294f0985c2 |
25-Jul-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix compilation on windows with clang, indentation cleanups
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6779 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/httpcommon.cc
ase/schanneladapter.cc
|
| 190d269c0f3a1857a11bb12d61c758361737b70a |
25-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
(This is a reland of the original attempt in r6747)
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6778 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 06b04ec4ab5f0366fa20b286588c63f74141ea11 |
24-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.
BUG=3579
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6776 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
|
| 45304ff0a712cf23d0de98d9e8f4fc576971b120 |
24-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71829282-> 71834788
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
xamples/call/callclient.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| 39f831fbb08992bf7abd0fd05d2af0fc1f8756d0 |
24-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-revert of 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.
BUG=N/A
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6772 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 437d57db5b3c7545d5edba2f123f8bcbdf2f80a3 |
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71775619-> 71778545
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 8c7e3291a956901f00aab813867cc21a35709d4d |
23-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.
BUG=N/A
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 8721f989bfd7e1c0a12e486383f90357564c4e78 |
23-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."
BUG=N/A
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6769 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.h
|
| e2da234e2767ad6e6433fbeee998b0f681100981 |
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71766184-> 71775619
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
xamples/call/callclient.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| 21b4da8ebd0fcdd981ad8a91e1100018a4192e1a |
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71753329-> 71766184
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6767 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| 0f7328cd6bf5d79c9c8ccf527ffef2827119e83b |
23-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Temporarily add a default ctor to StatsReport and make |id| non const.
As soon as I've updated the chrome side, I'll revert this cl.
TBR=henrike
Review URL: https://webrtc-codereview.appspot.com/16149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6766 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statstypes.h
|
| 9359cb3e75c7100dab4c687f60dd28dc613280e4 |
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable SendAndReceive tests.
Also fixes a crash in ::SetCapturer which wasn't exposed by tests
before.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6765 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 5ff71ab4b369fe3dbfaec5f91cd2e491397eff33 |
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "(Auto)update libjingle 71675033-> 71726409"
This reverts commit r6761 which looks like an accidental auto-revert of
r6760.
BUG=1788
TBR=wu@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6763 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 89c833cd9da0d1316928ff2235ca33e3e2117271 |
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71726409-> 71726772
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6762 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| f67f6aa74187dbb804ec3bc98b9551db9fcf5571 |
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71675033-> 71726409
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6761 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 8120353342f27df70018a808efa92acc8a07d9f2 |
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement suspend-below-min-bitrate option.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6760 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 543e589205af006f6b999a2c5df51d3fb722d925 |
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up VideoOptions for payload-based padding.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6759 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| efe4b9af49d2dbbf39a2f41b5818829a38fa0d5e |
22-Jul-2014 |
glaznev@webrtc.org <glaznev@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add VP8 video decoding hw acceleration support to Java Peerconnection library.
For now NVidia decoder is supported only,
Qualcomm will be added once b/16353967 is fixed.
TODO:
- Support queuing 2-3 decoder input buffers.
- Add average decoding time statistics.
- Add Qualcomm hw decoder support.
BUG=3030
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6758 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
ibjingle.gyp
|
| 6f48f1bf68a10669c9bcd81262c1a98ed2a8d462 |
22-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement encoder options in WebRtcVideoEngine2.
Implementing default options to enable denoising by default and wiring
up encoder settings to propagate VP8 settings.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6757 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| cadd078cf994b873f344621901fe62a621bbaa6c |
22-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused config.h and math.h includes.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6756 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 85f42949d66a86b382ef9ba9ec0fe496890bde08 |
22-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable ReceiveStreamReceivingByDefault test.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6754 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| fa5fcd671de2f6db6249c3cdd2908f4fc39d84a0 |
21-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71599033-> 71605904
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6751 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideocapturer.cc
|
| e69b0619266b9aab55feded823e4c44d7be51a8c |
21-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71575585-> 71599033
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
|
| 908f57ed945d7b8b47ec4cb50435a484cd6edf18 |
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable GetStatsForInvalidTrack while I rewrite it.
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17969005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6748 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface_unittest.cc
|
| 756b8462ebcd9d74234384239428d05f64907fa2 |
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6745
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| fd61a1d693840b3e177cade683f3e6d3d0119a9d |
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6745 "Refactor StatsCollector and associated types."
Broke build on android.
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/18819004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 647e05cfcdb5028a5556ec0268a65ea6794f47a8 |
21-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/mockpeerconnectionobservers.h
|
| 3c10758b3bb9519d5e582c00f454ac30196ac4e7 |
20-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check before send/receive rtp header extensions.
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13949004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 8fdeee6abfcb560233b5e769afb1c1c72cc2100d |
20-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement Base::ConstrainNewCodec2.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6743 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 3edbaaf33744e5b24e6946d4b84174e8db2d161e |
19-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Ignore empty data in DataChannel::Send to match FF's behavior.
BUG=crbug/395205
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6742 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
|
| 99f6308a2d2e4ef07bef63b2cf9f963d945af8a7 |
19-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71460499-> 71464449
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6741 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
|
| a0b929b63c35089a086715640149cdd24960fb2b |
19-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Reland r6707 with the fix for callclient.cc."
Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.
TBR=wu@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/17979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
xamples/call/callclient.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| 196ae6d667b090a9d36da5639ae4d9cb3cfd0ef2 |
18-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71456344-> 71456420
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6739 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
|
| 3dec81a736d402954f8043fb7636346cdd24198f |
18-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71456173-> 71456344
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
|
| a6e8cf8fb72b3c5bf331938ddb86093559c1c631 |
18-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reland r6707 with the fix for callclient.cc.
TBR=mallinath@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/13039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
xamples/call/callclient.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| 60e65b11c19a6021e5c1fbea461925b279b38b34 |
18-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71452608-> 71453580
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
|
| 8636fc852e0bc22a7f582d88e54211c858ec92aa |
18-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Creates the default track if the remote media content is send-only and there is no stream in the SDP.
BUG=2628
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling_unittest.cc
|
| e6f84ae8a602ce78733d20b280ce32198e7ecef5 |
18-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial WebRtcVideoEngine2::GetStats().
Also forward-declaring and moving WebRtcVideoRenderer out of header.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| d1ea06b3d5adab352741df5092c56b20f3e1a74f |
18-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Restart VideoReceiveStreams in WebRtcVideoEngine2.
Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.
Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| c31651d8474a1f7cdf134e19af54e98669a29089 |
18-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71378257-> 71410012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6726 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
|
| aa9361137586aef3e7bd2dc5625d3a7c91fd75da |
17-Jul-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Connect to the turn server if address cannot be resolved by the browser by using
unresolved address. This case is only considered for TCP sockets. P2P layer will
assume socket will do the resolve by using a proxy.
BUG=3384
R=jiayl@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6722 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
|
| e5995aadd5fb6c649cbe191e8c987df85b38b79c |
17-Jul-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.
This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.
BUG=3223
R=jiayl@webrtc.org, juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
2p/base/candidate.h
2p/base/p2ptransportchannel.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
|
| e10d28cf14d55a86138da97cbf87ca06bb2f5589 |
17-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
fix
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| 5301b0f1fce9478dfa56476e174332a1d67b053a |
17-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move additional state into WebRtcVideoSendStream.
Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/base/streamparams.cc
edia/base/streamparams.h
edia/base/streamparams_unittest.cc
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 52eddec71b45e8e5ff1294040b8cb658dd144c7a |
17-Jul-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6707 "Add support of multiple STUN servers in UDPPort."
Reason:
Breaks the build on callclient.cc.
> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
>
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
>
> BUG=3310
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13879004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| 4c3e9917e7431ba1c0d20535602209310ce48ded |
16-Jul-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
m= (media name and transport address)
i=* (media title)
c=* (connection information -- optional if included at
session level)
b=* (zero or more bandwidth information lines)
k=* (encryption key)
a=* (zero or more media attribute lines)
BUG=2260
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 46fb331bc5836eb03bc0cbda46097d9089a19561 |
16-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support of multiple STUN servers in UDPPort.
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
BUG=3310
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker.cc
2p/client/connectivitychecker.h
2p/client/connectivitychecker_unittest.cc
2p/client/httpportallocator.cc
2p/client/portallocator_unittest.cc
|
| a8d8ad2be6b7c204bbdc8c20a942e0aefb4fa347 |
16-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71240799-> 71250251
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtccommon.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 38ce7d03d83ec64c5394d425865803bf0894625f |
16-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement unittest for SetSendCodecsChangesExistingStreams.
BUG=1788
R=pbos@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19869004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6699 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 47218956fc432432146f9dadd7e266089ac94448 |
15-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Minor refactoring of StatsCollector.
* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.
The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
|
| 42fe4350fed39bcfe5e490a7b82c207862555f2e |
15-Jul-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove Thread::RunningForChannelManager().
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.
BUG=3388
R=andrew@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread.h
ession/media/channelmanager.cc
ession/media/channelmanager_unittest.cc
|
| 2adc51c86e075c3a1f396fdcfad68f974f5adf57 |
15-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Handle the case if an unusually long peer name is provided in the peerconnection example.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6687 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/server/peer_channel.cc
|
| cb859ecd3b9435633434ca3c028eb60c8e8c5938 |
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace strcpy with talk_base::strcpyn.
Cpplint reports error 'Almost always, snprintf is better than strcpy'
when checking code styles. The function talk_base::strcpyn() is a better
option than strcpy().
BUG=1788
R=pbos@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12919004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6686 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 1b84116417d5b5809482cd0b0d9dd4af54668508 |
14-Jul-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a facility to the Thread class to catch blocking regressions.
This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.
This is a reland of an already reviewed cl (r6679) that got reverted by mistake.
TBR=xians@google.com,tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6682 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread.cc
ase/thread.h
|
| b038c723696ecb00dde71f7b4ba626265cd7d4c2 |
14-Jul-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable SCTP compile for iOS.
Chromium's been updated to pull a version of usrsctplib that will compile correctly. This update DEPS to point at new revision and turn on the compile time flags for iOS sctp.
BUG=3211
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6681 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
ibjingle.gyp
|
| aac14973aa9b9633683f6dd791983734b8ba959c |
14-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71116846-> 71117224
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6680 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread.cc
ase/thread.h
|
| 5be649fcfce6f8ef66c743894742cbd4fbc95122 |
14-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a facility to the Thread class to catch blocking regressions.
This facility should be used in methods that run on known threads
(e.g. signaling, worker) and do not have blocking thread syncronization
operations via the Thread class such as Invoke, Sleep, etc.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/12999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6679 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread.cc
ase/thread.h
|
| 242068d58cc01640aa9f733fa67f078fc65c4ae5 |
14-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
A step towards changing StatsReport::Value::name to an enum.
The stats reporting code does a lot of unnecessary string copying.
This is a step in the direction of removing that and forcing use of only known constants.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/12989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6678 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
|
| 03505bcb7a369add7abfe306004e7803ab096f21 |
14-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make StatsCollector depend on always having a valid session pointer.
This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it.
I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/13959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6677 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
|
| b5348c64bb3d319ecdfe096cb4fb5fcecf38f838 |
14-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Minor refactoring of the session classes.
Make member variables that never change and are touched on multiple threads, const.
Move implementations of setters/getters of variables that can change, into the cc file in preparation of adding thread correctness checks.
This is a relanding of a cl already reviewed but got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/12979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6676 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/session.cc
2p/base/session.h
ession/media/channel_unittest.cc
|
| d8524348bbb9e5b960f670d84cb689c46f49b3de |
14-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71107853-> 71115715
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| b92f6f93716ee9f84795bfb79d747dd982c75d03 |
14-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71099685-> 71107853
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6674 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 5f43ce6784b64810e2431343fd3d90d431e37cc1 |
14-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a type cast issue for compiling webrtc with BoringSSL.
BUG=
R=juberti@google.com, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6672 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/openssladapter.cc
|
| e04cb0eb81d8b99f87b5e3c675b8583b7a9edd63 |
14-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70948025-> 70959275
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6671 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/logging.h
|
| ccbed3b3c4a0f7607eadafd2c1edb7578d32f099 |
11-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement unittest SetRecvCodecsAcceptDefaultCodecs.
BUG=1788
R=pbos@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14869004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6663 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 72670206dba08238a4fd231867366fe67c37b3a3 |
09-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70813271-> 70818369
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6642 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 4b1f330b4fc07066e028be782655185f229216de |
09-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a bug in SocketAddress where "a.b.c.d:1" and "b.b.c.d:1" are incorrectly considered equal.
BUG=3558
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6639 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/socketaddress.cc
ase/socketaddress_unittest.cc
|
| e9cefdef68eb42caf6f249ae5d99dfd0f5ebdaa0 |
09-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improve libjingle's ASSERT and VERIFY macros on Windows.
This change has the effect that when using a debugger, a failing ASSERT/VERIFY will break exactly where the failing expression is and not two callstacks up.
Minidumps (for debug builds) will also have the failing expression at the top of the call stack.
R=xians@webrtc.org, xians
Review URL: https://webrtc-codereview.appspot.com/12929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6633 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/common.h
|
| 01bda2068bebb65a610c0d951f938db5dd028394 |
09-Jul-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed the stats problem when new track is using the same ssrc as the previous track.
Before this patch, when switching from voice mode to stereo mode, the stats won't be updated because StatsCollector binded the ssrc report with the old track, so the report can't be updated by the new track.
This patch fixes the porblem by changing the ssrc report track id to use the new track id.
TEST=libjingle_peerconnection_unittest --gtest_filter="*StatsCollectorTest*"
R=hta@chromium.org
Review URL: https://webrtc-codereview.appspot.com/17859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6632 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
|
| 55535d4e5802689d554a6c1a90f0154bb5c64b3c |
08-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70711261-> 70733822
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6627 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
ibjingle_tests.gyp
|
| ecb8723402f2c138466ff9dc2ecc8626b7fb3db5 |
08-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change Timing::WallTimeNow to be static.
There's no need to construct a Timing object to call this method.
On Windows we were unnecessarily calling CreateWaitableTimer + CloseHandle but never actually using that waitable timer.
There's otherwise no change in functionality.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6624 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
ase/timing.cc
ase/timing.h
|
| a70be68f651a75930681b7607fd4da054de68842 |
07-Jul-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling shared socket mode for TURN ports. This is done as currently when
TURN server also used as STUN server, binding responses will be handed over
to TURN port, which simply discard these messages, as requests are originated
from StunPort.
Until we find the right solution for this problem, it's better we disable this
feature.
BUG=https://code.google.com/p/webrtc/issues/detail?id=3537
R=jiayl@webrtc.org, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6618 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/portallocator.h
2p/client/basicportallocator.cc
|
| bd249bc711b3c9efd142eb8de3df489282fe693e |
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove GetDefaultConfigs() from Call.
Defaults for configs are instead placed in the Config constructors.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 3ffa1f917ec1a8bd7666669ddb3f8ba0fd26cb4e |
02-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70422491-> 70424781
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6586 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 0bb9fac98ca95509e7c07debaee316bdaa2f4eaa |
02-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70343444-> 70394475
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6581 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| d8a90690809f0fa57e88911fb96848e227947424 |
01-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70340027-> 70343444
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6579 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 74bf7a65238cebfba51377d0d81e4a58f097c1ff |
01-Jul-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add tkchin@ to OWNERS.
Adding myself to OWNERS of subdirectories containing iOS bits. Added niklas.enbom@ for audio_device and wu@ for everything else.
R=niklas.enbom@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/OWNERS
|
| 974bbbb352f61f7e7c0d959858a9c471ce0f26f0 |
01-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix uninitialized value in DtlsTransport and TransportDescription.
BUG=crbug/390304
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6577 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/dtlstransport.h
2p/base/transportdescription.h
|
| 633564540036e1e4bb00308911edbfb303f51fe6 |
01-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70329914-> 70330023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6575 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 0402515d35d78717f4a02ef6daed8036127f48aa |
01-Jul-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement command line flags for peerconnection client example on Windows
Adding the flags and functionality for 'autoconnect', 'autocall', 'server',
'port', and 'help' like in the linux example.
BUG=3459
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13609004
Patch from Vicken Simonian <vsimon@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6573 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/client/main.cc
xamples/peerconnection/client/main_wnd.cc
xamples/peerconnection/client/main_wnd.h
|
| d5a0506e847e55d8f9145d5e548e98302f264e22 |
30-Jun-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use X509_NAME, not struct X509_name_st.
Also include openssl/x509.h explicitly since we're using functions and types
from it.
BUG=none
R=henrike@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6569 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/openssladapter.cc
|
| bfa758a54c066f4d0bb125e102fbb654ee177a88 |
27-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 70004190-> 70103367
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6555 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
|
| 269605ce450545c565a77719e0167024a7d3643c |
26-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement SetSendCodecs() unit tests for WebRtcVideoChannel2.
BUG=
R=pbos@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12829004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6543 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 420ca434b15c63d6a9491111d5adbfbeaf57afb4 |
26-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69860953-> 70002228
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6542 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| ec9f5fb34cc612f26ec30f357ea3e3aa5d96c5c2 |
24-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE].
BUG=crbug/387632
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/17779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6533 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| 9a4f651037918e83bb0cf4c25e4910551b19659c |
24-Jun-2014 |
aluebs@webrtc.org <aluebs@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 for TSAN2
BUG=webrtc:3498
R=henrik.lundin@webrtc.org
TBR=tommi
Review URL: https://webrtc-codereview.appspot.com/21689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6528 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/physicalsocketserver_unittest.cc
|
| 71dffb76dcc59fcd886c4899b91a7a48db6ea254 |
24-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69648312-> 69830415
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6527 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/base/videoadapter.h
edia/webrtc/webrtcvideoengine.cc
|
| ff1b1bf0944d42700edadae68bd774835a7a13f0 |
20-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
When creating an answer, takes the codec preference from the offer.
This change is based on RFC3264:
"Although the answerer MAY list the formats in their desired order of preference, it is RECOMMENDED that unless there is a specific reason, the answerer list formats in the same relative order they were present in the offer."
BUG=2868
TEST=unit tests and manually with munge-sdp test
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/14589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6514 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/codec.cc
edia/base/codec.h
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient_unittest.cc
|
| 0d15159b041f34855a291322d6a785211244e02d |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69634309-> 69640360
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6512 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcexport.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| b43c99de297e2686233cf495625ba1d87cbfe0e4 |
20-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Limits the send and receive buffer by bytes, not by packets.
The new limit is 16MB for each buffer.
Also refactors the code to handle send failure more consistently.
BUG=3429
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6511 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
|
| db397e5c6c387ffb108f71059cb993e25c47a6fc |
20-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-evalutes the ICE role on ICE restart.
Also unifies the logic of ICE restart.
BUG=1775
R=juberti@google.com, juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6510 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
2p/base/p2ptransportchannel.cc
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
|
| bb2d65895b14d5ab6282144f2eb223f134c7f74d |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69617317-> 69623266
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6508 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
ibjingle.gyp
edia/base/mediaengine.cc
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 75ce92086c955d7cba7d4fc9ffaba80097ce178c |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69600065-> 69617317
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6507 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel.cc
ession/media/channel.h
|
| 83785d37d119fc323abe41609052edc149c74197 |
20-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused ALLOCATE_DELAY constant.
Breaks linux_tsan2 compile [-Wunused-const-variable].
TBR=mallinath@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/20749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6505 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
|
| 4c25c671466b56aecd7581ea86342746d274a6bd |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69589535-> 69600065
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6504 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
|
| 58e7c8660c1b3a26ec4901d3d763279069bf7057 |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69588980-> 69589535
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6503 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
ibjingle.gyp
edia/base/mediaengine.cc
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcmediaengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 0970dd8767d91be5b0872c3210e93ed355107b71 |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69588608-> 69588980
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6502 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
|
| 8563ef448a9dcf7cd5755da488b29e7a7f9cc5de |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69587333-> 69588608
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcexport.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 1ef789d455700af127b788a9befbe77075bd29c3 |
20-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69568113-> 69587333
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6500 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/rtputils.cc
edia/base/rtputils.h
ession/media/bundlefilter.cc
ession/media/bundlefilter_unittest.cc
ession/media/channel.cc
|
| df9bbbee56f4d9ecef93b3c46964b6f29803f81b |
19-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69567902-> 69568113
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6498 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/constants.h
edia/webrtc/constants.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine2.cc
|
| fbd13286dc280eaa69c562e20e11a38cb393da3d |
19-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69555283-> 69567902
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6497 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.h
edia/base/codec.cc
edia/base/codec.h
edia/base/codec_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
|
| 21794f9862bc55288f7ca5098cac89fc108c680c |
19-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69543894-> 69555283
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6496 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| d27d9ae644c20c91ca6064bc17ffe2cca0f1be2c |
19-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69506154-> 69515138
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6488 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| acede34aea92cb07049e187341a132f92a34662a |
19-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a memory leak in SctpDataMediaChannelTest.
BUG=3492
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6486 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine_unittest.cc
|
| f8063d34deefb55b4a0e5091fc59d5d5e58e43d8 |
18-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Properly shut down the SCTP stack.
TBR phoglund@webrtc.org for the tsan_v2/suppressions.txt change.
R=ldixon@webrtc.org, pthatcher@webrtc.org
TBR=phoglund@webrtc.org
BUG=2749
Review URL: https://webrtc-codereview.appspot.com/12739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6484 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
ibjingle_tests.gyp
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
|
| 2eaac188bbda9fb2b838a71833024d1975360fa1 |
17-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Makes the sid of a closed DataChannel available to reuse per the spec.
BUG=2646
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6468 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/webrtcsession.cc
ase/common.cc
ase/common.h
|
| ed3e0d8f0d5b277c298eedd246cbe93762443edf |
17-Jun-2014 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Increasing tolerances quite a bit to fight flakes.
From these errors:
[----------] 3 tests from ProfilerTest
[ RUN ] ProfilerTest.TestFunction
../../talk/base/profiler_unittest.cc:56: Failure
The difference between kWaitSec and event->mean() is 0.13612610600000002, which exceeds kTolerance, where
kWaitSec evaluates to 0.25,
event->mean() evaluates to 0.38612610600000002, and
kTolerance evaluates to 0.10000000000000001.
[ FAILED ] ProfilerTest.TestFunction (655 ms)
[ RUN ] ProfilerTest.TestScopedEvents
../../talk/base/profiler_unittest.cc:98: Failure
The difference between kEvent2WaitSec and event2->mean() is 0.33170768900000003, which exceeds kTolerance, where
kEvent2WaitSec evaluates to 0.14999999999999999,
event2->mean() evaluates to 0.48170768899999999, and
kTolerance evaluates to 0.10000000000000001.
I didn't spend time understanding why; I reckon the test had too tight
tolerances to start with so I'm just adjusting them a bit. That's
probably better than disabling the test, now it still has some value.
R=aluebs@webrtc.org
TBR=aluebs@webrtc.org
BUG=None
Review URL: https://webrtc-codereview.appspot.com/13729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6464 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/profiler_unittest.cc
|
| ae740dd94cb4f11271e5dc9b27eee1f2e29a37a8 |
17-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69359922-> 69365993
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6463 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 44a317a6983402c63db8b3cd44f69efc7245b815 |
17-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69337301-> 69359922
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6457 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| 53f57936c1fbe0caaabce7ccb85b77935fd97fa8 |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69306183-> 69323802
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6454 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer_unittest.cc
|
| 587ef60056ff0e301a95a9eb8231fb0cae6b69b1 |
16-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement RTP extension support in WebRtcVideoEngine2.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6453 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| d054bff3b9a23ddf1e8c0c844f13bc4b10540689 |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69292418-> 69293749
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6452 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 88d9fa63df0dc703545197a61854be0e9fb1f6a4 |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69291002-> 69292418
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6450 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
|
| 27626a6256878611fd2dd10a4e6e1c464fd79463 |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69278008-> 69291002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6448 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 0a1e7e0b004628344e95f0300e7de6bb8418594a |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69276003-> 69278008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6442 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| d1591409658e3b35f734dd1b0026661d01c796b5 |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69260070-> 69276003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6439 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 117afeec910a01481000c22b46c66c5ddb9f8e4e |
16-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69188577-> 69260070
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6437 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.h
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor.h
|
| ab23d493e0f0c2be0abf2c55770e242e37d97a1e |
14-Jun-2014 |
glaznev@webrtc.org <glaznev@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.
Review URL: https://webrtc-codereview.appspot.com/20659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6436 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/OWNERS
|
| c6c1dfd7ea5ff73c5ea224719c27a09083b9d7f0 |
14-Jun-2014 |
glaznev@webrtc.org <glaznev@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add extra logging and latency restriction to VP8 HW encoder.
- Do not allow encoder to accumulate more than 100 ms of
data in input buffers.
- Add optional extra logging (disabled by default) to track
encoder buffers timing.
BUG=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6435 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| a6764ab8699eae79825f716fa281c3495bc9ad3d |
13-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69144530-> 69164179
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6434 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| db56390f7e6a1bce80cc49635f039f225679860f |
13-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69143161-> 69144530
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6432 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
|
| f99c2f2dbcaab24b45295cb9e06c3c52ad349d81 |
13-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add NACK feedback parameter to WebRtcVideoEngine2.
Also fixing enabling/disabling of NACK. Previous implementation was made
under the assumption that NACK should always be enabled which caused
both missing NACK settings in SDP as well as broken interop between old
and new WebRtcVideoEngines.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6431 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| e322a175f6f38c4ed39296d9724edf005e536a63 |
13-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement RTX tests+fixes in WebRtcVideoEngine2.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6430 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 9fbb717acaf7c9914ad145d72511efc5135ab248 |
13-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove engine_codecs_ cache from unittests.
Used interchangably with engine_.codecs() becomes confusing and it's not
really used that much.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6429 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| d54ec1256c06f1fe9fb86f3dc5940c6f06a47f5e |
13-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix GYP DEPTH for libjingle isolate files
In https://review.webrtc.org/13679004/ the libjingle isolate
files in patch set #2 were not tested, which caused a failure when
6427 was committed. This fixes the talk/build/isolate.gypi with a
similar change.
BUG=343106
TEST=Successful local compile on Linux
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6428 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
|
| a1bfc50a725434ccb45f65e68ede5ea0085738da |
13-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Pass GYP DEPTH variable to isolate.
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.
Also update all our .isolate files to use the <(DEPTH)
variable.
BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.
R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_unittest.isolate
|
| c800c1cc4080275f81ea5378d2edeaad04564bc0 |
13-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69131548-> 69132244
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6426 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
|
| 1c8223c590c154ae99b04a3a55e2bb459afb7185 |
13-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial owners file for talk/media/webrtc/.
Including pthatcher@webrtc.org (already root owner) and
mflodman@webrtc.org.
BUG=
R=juberti@google.com, juberti@webrtc.org
TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15679008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6425 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/OWNERS
|
| 7e71b77f8aab5b7a6f2b669c16f90ec9a4b4609c |
13-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69102234-> 69116997
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 1a6c6281ca028e4fbba5c015ed7166ffc34bae9c |
12-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck'
Failing tests are disabled for memcheck.
TBR=wu@webrtc.org
BUG=2626
Review URL: https://webrtc-codereview.appspot.com/13699004
Review URL: https://webrtc-codereview.appspot.com/13699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6422 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
edia/sctp/sctpdataengine.cc
|
| ddeec048c093da6e8e3dc17e599672681fa4def7 |
12-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck
This reverts commit c3272a942f04f9dd0db3f6bf0d201bcf47c3fa08.
TBR=wu@webrtc.org
BUG=2626
Review URL: https://webrtc-codereview.appspot.com/13689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6420 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
edia/sctp/sctpdataengine.cc
|
| 3f3f428d2ba733b78368034c46da0653ba867ef6 |
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69097619-> 69099564
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6419 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 6c6f33b5bb934602896cdf06c9397fca1b9f6bdf |
12-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the flaky RTP DataChannel test.
BUG=2891
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6418 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 18dfa8d5741443bc0a8a3e99b821516aa28ced01 |
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69069003-> 69082899
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6417 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 4cb012858f7461015e405c0c2cfc4b9f10a086ce |
12-Jun-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixed GetStats when local and remote track are using the same ssrc.
R=hta@chromium.org, kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6414 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastream_unittest.cc
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ibjingle_tests.gyp
|
| b90619c07fb9b9723ad5160651ab416724d3fa61 |
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69049090-> 69054765
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6412 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| d41eaeb7cded2b2cda83f53aa320cf18e2d07380 |
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69005149-> 69049090
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6408 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
|
| e9e8007ab4b5bf29b0590e2cf0cdbc358c41dcc6 |
11-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68985065-> 69005149
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6406 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 9e65a3b0135467d0508875d675f0551e9c7fe82a |
11-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-land webrtcmediaengine.cc part of r6397.
webrtcvideoengine.cc un-reverted by a bot roll in r6399 so half of r6397
is still applied. The applied fix (diff between r6397) is to put
WebRtcVideoEngine2 in ifdefs and only build for WEBRTC_CHROMIUM_BUILDs
corresponding to webrtcmediaengine.h.
BUG=
R=minyue@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19719005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6401 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
|
| 5d223a7d2d83206e9061071af59b670c9a7687e2 |
11-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68982444-> 68983526
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6399 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 6604c6df260cc31dcc57b2a6dc3bb476b6526f40 |
11-Jun-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6397 "(Auto)update libjingle 68949184-> 68982444"
> (Auto)update libjingle 68949184-> 68982444
TBR=buildbot@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6398 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideoengine.cc
|
| af214d804fe72bb7532081f4eb63c1a21ce74a88 |
11-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68949184-> 68982444
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6397 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.cc
edia/webrtc/webrtcvideoengine.cc
|
| e61b8e32d8c670f66508f2316a4215ef97bd0ab8 |
11-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds end to end DataChannel tests.
BUG=2626
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6390 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
edia/sctp/sctpdataengine.cc
|
| a40210aee29cb9b682edbdad2a11df0878673170 |
11-Jun-2014 |
glaznev@webrtc.org <glaznev@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support for NVidia VP8 HW encoder.
- Some changes in HW VP8 encoder search logic to detect HW codec
with supported color space format.
- Support yuv420 and nv12 formants for encoder input.
- Add some extra logging and encoder frame drop statistics.
BUG=3176
R=fischman@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6389 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| 1014101470940fc60445c1573a3da14784f63b0e |
10-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6380 "Replace libjingle_root with talk_root variable."
It turns out this doesn't fix the problem we're trying to solve...
> Replace libjingle_root with talk_root variable.
>
> This CL is similar to https://review.webrtc.org/9019004/
> It is needed in order to be able to build with different
> copies of libjingle. Having the libjingle_root variable didn't
> make this possible, since relative paths in the .isolate files
> ended up at the wrong directory level and .isolate files doesn't
> support all the normal GYP variables like <(DEPTH).
>
> BUG=chromium:343106
> TEST=trybots passing compile step with clobber.
> R=tommi@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15709004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6384 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| 3eb2c2f4c35ab62ede70d34699eecc17956b0fcf |
10-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68891947-> 68893961
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6383 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 86f613d6b893d03c822373f7e7ec51db78a90f9f |
10-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move WebRtcVideoEngine2 fakes to unittest header.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6382 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
edia/webrtc/webrtcvideoengine2_unittest.cc
edia/webrtc/webrtcvideoengine2_unittest.h
|
| 02386829842bd38d70cb3016227b78c46c06620e |
10-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace libjingle_root with talk_root variable.
This CL is similar to https://review.webrtc.org/9019004/
It is needed in order to be able to build with different
copies of libjingle. Having the libjingle_root variable didn't
make this possible, since relative paths in the .isolate files
ended up at the wrong directory level and .isolate files doesn't
support all the normal GYP variables like <(DEPTH).
BUG=chromium:343106
TEST=trybots passing compile step with clobber.
R=tommi@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6380 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| 6b6e58d6325067977770ca7dfe4bef8457dd0141 |
09-Jun-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused test_env.py from isolate files + fix nss path.
This is not necessary for executing tests for WebRTC.
It probably appeared in our .isolate files because of code
copied from Chromium.
BUG=
TEST=All non-baremetal trybots passing.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6373 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_unittest.isolate
|
| 85d2794e5b57a1501a7fdade61eccd086e7a622d |
09-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for the "apt" format parameter and turns on the RTX feature.
BUG=1811,1095
R=henrike@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6372 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| e3cdd9959e5ff1dc2c9850aeedb6b8671d3eb0f9 |
07-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio."
This reverts commit 56631a14bdae24aa0bfaceeb2b57df729fee1227.
TBR=henrike@webrtc.org
BUG=3235
Review URL: https://webrtc-codereview.appspot.com/19669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6363 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/rtputils.cc
edia/base/rtputils.h
2p/base/dtlstransportchannel.cc
ession/media/bundlefilter.cc
ession/media/bundlefilter_unittest.cc
|
| 013bdf802a613d54fb8c234604185cedddb73e9b |
07-Jun-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
APPRTCDemo(objc): Remove regex parsing in favor of JSON struct.
Also some cleanup/refactoring of APPRTCAppClient.
R=fischman@webrtc.org, glaznev@webrtc.org
BUG=3407
Review URL: https://webrtc-codereview.appspot.com/18499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6362 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/objc/AppRTCDemo/APPRTCAppClient.h
xamples/objc/AppRTCDemo/APPRTCAppClient.m
xamples/objc/AppRTCDemo/GAEChannelClient.h
xamples/objc/AppRTCDemo/GAEChannelClient.m
|
| c3288c130d34e009be91c6d477989d523a090fbd |
06-Jun-2014 |
glaznev@webrtc.org <glaznev@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add OpenGL Android video renderer which can display multiple
yuv420 images in a single GLSurfaceView.
Start using new video renderer in AppRTC demo app.
BUG=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/android/org/webrtc/VideoRendererGui.java
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/AppRTCGLView.java
xamples/android/src/org/appspot/apprtc/FramePool.java
xamples/android/src/org/appspot/apprtc/VideoStreamsView.java
ibjingle.gyp
ibjingle_examples.gyp
|
| 745a39cced5e5fc5ed9d1c2df2d4659f0470ad8a |
06-Jun-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio.
BUG=3235
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6356 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/rtputils.cc
edia/base/rtputils.h
2p/base/dtlstransportchannel.cc
ession/media/bundlefilter.cc
ession/media/bundlefilter_unittest.cc
|
| 9512719569b86f0cad069a2fc1ce4bbc06eba974 |
06-Jun-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): support app (UI) & capture rotation.
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.
BUG=2432
R=glaznev@webrtc.org, henrike@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
xamples/android/AndroidManifest.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/VideoStreamsView.java
|
| 91c910469fa97ae16bd4c1e456cdeb2f0bf43faa |
06-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68701339-> 68703656
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6352 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 910473b31aa0f9c48aeba269c28ea632e0f06b12 |
06-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix C++11 -Wnarrowing in channel_unittest.cc.
Implicit conversion from int to unsigned char inside {} initializers is
ill-formed C++11 and triggers a warning in clang when building it as
such.
BUG=
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6351 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel_unittest.cc
|
| 7b6cbb3aa035414a36d3b6a0526c735502103763 |
06-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68689052-> 68689059
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6350 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 6ae48c660934784b4df56ab1ac99402ce3745e9f |
06-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VideoSendStream/VideoReceiveStream configs const.
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.
CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.
This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260
Review URL: https://webrtc-codereview.appspot.com/20409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 4b83a471defbdb42148bada873cfb66082191727 |
05-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68646004-> 68648993
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 94454b71adc37e15fd3f5a5fc432063f05caabcb |
05-Jun-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
|
| 130fa64d4c726765c66879e440e27e7bda86508f |
05-Jun-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct.
BUG=3407
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6345 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
|
| 0d523eea831e616c415c61765127ed5eb17e5f11 |
05-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove static initializer from WebRtcVideoEngine2.
BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| f1adbeedb4b4a56cfeb3e0789c2bb900762ec977 |
04-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68562943-> 68571194
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6333 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/genericslot.h
ase/genericslot.h.pump
ase/genericslot_unittest.cc
ase/sigslottester.h
ase/sigslottester.h.pump
ase/sigslottester_unittest.cc
|
| 738df8913db644d33c717b11b9155a2e10e3c6cf |
04-Jun-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix retain cycle in RTCEAGLVideoView.
CADisplayLink increases its target's refcount. In order to break retain cycle, we wrap CADisplayLink in a new RTCDisplayLinkTimer class and use that instead.
R=fischman@webrtc.org, noahric@chromium.org
BUG=3391
Review URL: https://webrtc-codereview.appspot.com/16599006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6331 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCEAGLVideoView.m
xamples/objc/AppRTCDemo/APPRTCConnectionManager.m
|
| 6f237769b3d74b10a138184731c3fef2130bec0b |
04-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68507189-> 68543735
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6329 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 40b45fc07a214c4102d83883bbd8bb7521de11e2 |
04-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68506654-> 68507189
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6328 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 0cdcd23a03ae78eb12bfbba9d71df7ef05e09448 |
04-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68501302-> 68506654
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6321 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/webrtc/webrtcvideoengine.cc
|
| af81b9bffd7ce21ea476b11748ebd7c14af5a117 |
04-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68499439-> 68501302
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6320 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 251fdf64cbd0c41adc65320e0c08cc73037f9e7f |
04-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68495561-> 68499439
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6319 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
|
| 09a71cd9ce0f655a29ccac0bb285155ce6c9a4e9 |
04-Jun-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291).
BUG=N/A
R=tkchin@webrtc.org
TBR=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6318 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/AppRTCDemo-Prefix.pch
xamples/ios/AppRTCDemo/Default.png
xamples/ios/AppRTCDemo/GAEChannelClient.h
xamples/ios/AppRTCDemo/GAEChannelClient.m
xamples/ios/AppRTCDemo/Info.plist
xamples/ios/AppRTCDemo/ResourceRules.plist
xamples/ios/AppRTCDemo/en.lproj/APPRTCViewController.xib
xamples/ios/AppRTCDemo/ios_channel.html
xamples/ios/AppRTCDemo/main.m
xamples/ios/Icon.png
xamples/ios/README
xamples/objc/AppRTCDemo/ios/Default.png
xamples/objc/Icon.png
|
| 53217848b28e1bc436cb1057df680b525e007815 |
03-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68465410-> 68487517
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6317 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
|
| 83eb7dff5ca37c35ed096ce3ab7b70db2610335a |
03-Jun-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED.
This should be reverted when COMPLETED is delivered reliably.
BUG=3021
TESTED=without this patch the test fails in Debug mode after a handful of runs. With this patch 100 runs passed in a row on my desktop.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6315 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
|
| 289a35c56dd5cbab7878ebb53030cd1f3d4c020f |
03-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add empty webrtcmediaengine.cc.
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.
BUG=1788
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/webrtc/webrtcmediaengine.cc
|
| b525a9d790b3fd5ec63aed92395623c3acdfd5b6 |
03-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68379861-> 68445177
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
|
| 044bdacfefa860715e84663d4df651e8f4984469 |
03-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove kMaxWaitForStatsMs from tsanv2 compilation.
As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.
BUG=1205,3220
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 34a08b4fb8fc36c08da215666840661ec86b58a7 |
02-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68275107-> 68379861
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/AppRTCDemo-Prefix.pch
xamples/ios/AppRTCDemo/Default.png
xamples/ios/AppRTCDemo/GAEChannelClient.h
xamples/ios/AppRTCDemo/GAEChannelClient.m
xamples/ios/AppRTCDemo/Info.plist
xamples/ios/AppRTCDemo/ResourceRules.plist
xamples/ios/AppRTCDemo/en.lproj/APPRTCViewController.xib
xamples/ios/AppRTCDemo/ios_channel.html
xamples/ios/AppRTCDemo/main.m
xamples/ios/Icon.png
xamples/ios/README
xamples/objc/AppRTCDemo/ios/Default.png
xamples/objc/Icon.png
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
2p/base/relayserver.cc
|
| 174a67439b03cc9c98bcc7fb426ddda8855a0fc2 |
02-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang.
Also removes one case of unused-variable.
BUG=3220
R=henrike@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6297 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
uild/common.gypi
|
| 8a09af3f673b8d1e4975b094a1d42c8092bb8fe2 |
31-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/17639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6296 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/opensslstreamadapter.cc
|
| 0163674f99681f5eb9c933545dacc8c04f140b4f |
31-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates.
It used to compre a parent certificate's digest against the SDP fingerprint and caused connection failure.
BUG=3383
R=bemasc@webrtc.org, juberti@webrtc.org, rsleevi@chromium.org
Review URL: https://webrtc-codereview.appspot.com/17589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6294 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/opensslstreamadapter.cc
|
| 56d114627b7ab2265939f95efba41572d3a1e6bb |
31-May-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix AppRTC target configuration in libjingle_examples.gyp.
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/18489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| acca675bcf8fefd1d1985ea3d0fb8f9ea65f5d4a |
31-May-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement mac version of AppRTCDemo.
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.
BUG=2168
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/README
pp/webrtc/objc/RTCEAGLVideoRenderer.mm
pp/webrtc/objc/RTCEAGLVideoView.m
pp/webrtc/objc/RTCNSGLVideoView.m
pp/webrtc/objc/RTCOpenGLVideoRenderer.mm
pp/webrtc/objc/RTCPeerConnection+Internal.h
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/public/RTCEAGLVideoRenderer.h
pp/webrtc/objc/public/RTCEAGLVideoView.h
pp/webrtc/objc/public/RTCNSGLVideoView.h
pp/webrtc/objc/public/RTCOpenGLVideoRenderer.h
pp/webrtc/objc/public/RTCPeerConnection.h
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/AppRTCDemo-Prefix.pch
xamples/ios/AppRTCDemo/Default.png
xamples/ios/AppRTCDemo/GAEChannelClient.h
xamples/ios/AppRTCDemo/GAEChannelClient.m
xamples/ios/AppRTCDemo/Info.plist
xamples/ios/AppRTCDemo/ResourceRules.plist
xamples/ios/AppRTCDemo/en.lproj/APPRTCViewController.xib
xamples/ios/AppRTCDemo/ios_channel.html
xamples/ios/AppRTCDemo/main.m
xamples/ios/Icon.png
xamples/ios/README
xamples/objc/AppRTCDemo/APPRTCAppClient.h
xamples/objc/AppRTCDemo/APPRTCAppClient.m
xamples/objc/AppRTCDemo/APPRTCConnectionManager.h
xamples/objc/AppRTCDemo/APPRTCConnectionManager.m
xamples/objc/AppRTCDemo/GAEChannelClient.h
xamples/objc/AppRTCDemo/GAEChannelClient.m
xamples/objc/AppRTCDemo/channel.html
xamples/objc/AppRTCDemo/ios/APPRTCAppDelegate.h
xamples/objc/AppRTCDemo/ios/APPRTCAppDelegate.m
xamples/objc/AppRTCDemo/ios/APPRTCViewController.h
xamples/objc/AppRTCDemo/ios/APPRTCViewController.m
xamples/objc/AppRTCDemo/ios/AppRTCDemo-Prefix.pch
xamples/objc/AppRTCDemo/ios/Default.png
xamples/objc/AppRTCDemo/ios/Info.plist
xamples/objc/AppRTCDemo/ios/ResourceRules.plist
xamples/objc/AppRTCDemo/ios/en.lproj/APPRTCViewController.xib
xamples/objc/AppRTCDemo/ios/main.m
xamples/objc/AppRTCDemo/mac/APPRTCAppDelegate.h
xamples/objc/AppRTCDemo/mac/APPRTCAppDelegate.m
xamples/objc/AppRTCDemo/mac/APPRTCViewController.h
xamples/objc/AppRTCDemo/mac/APPRTCViewController.m
xamples/objc/AppRTCDemo/mac/Info.plist
xamples/objc/AppRTCDemo/mac/main.m
xamples/objc/Icon.png
xamples/objc/README
ibjingle.gyp
ibjingle_examples.gyp
|
| 9f8164c06054016978378b7a01a9180106d92771 |
30-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix two bugs in DataChannel state transition.
1. OnStateChange should not be fired if state is not changed.
2. RemotePeerRequestClose should be a no-op if it's already closed.
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/21559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6290 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
|
| 1678db9df6d3af8db0ee3fd2018a75d1528bbc9b |
30-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68230113-> 68244456
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6287 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 540a2251aae599e23c0c34792b9794a537923ac7 |
30-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68230011-> 68230113
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6281 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 35efb839ed4593414f0cce55bb1d44b9fd9d59de |
30-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement new-API test RecvStreamWithoutRtx.
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/20449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6280 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| c34bb3a88627672d99b1c037d36dbeb23407fae4 |
30-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Log default receive stream creation.
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/17459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 198647473ba207d59dc94216ef38496d43d15592 |
30-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement and fix new-API NackIsEnabled test.
Required enabling NACK on receiver side which was apparently missed.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 1d66be22c8f929e1170f288472aac9d4b44b7a05 |
30-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68203780-> 68206793
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 8dcd43c4f71da88f75ca46ed5868eb8812e1d6f7 |
30-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.
BUG=2796
R=juberti@webrtc.org, pthatcher@google.com
Review URL: https://webrtc-codereview.appspot.com/13439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| abe01dd634cd92245b7836e687de0f6e7c0723b9 |
29-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): run in full-screen & immersive mode.
Also:
- Only show stats HUD on demand
- Only collect stats when HUD is showing
- Don't render solid green frame when video is not present in either direction
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/VideoStreamsView.java
|
| 5dc51fbe50e632d5db225a2f6cbaaba1700e976c |
29-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Closes the DataChannel when the send buffer is full or on transport errors.
As stated in the spec.
BUG=2645
R=pthatcher@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6270 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
ibjingle_tests.gyp
|
| 001fd2d5037cca62b717619827cde675ee35f470 |
29-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fire OnRenegotiationNeeded only for the first SCTP DataChannel.
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.
BUG=2431
R=pthatcher@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface_unittest.cc
|
| 43a13953708ee5081fcbc1255cb48ca62104b899 |
28-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): README updates for a shrinking envsetup.sh world.
There was duplicated (and out of date!) information in README relative to
getting-started so de-duped to point to getting-started as the canonical
reference.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15589006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6265 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/README
|
| b364016cbb7cf8f7050932bcd2b2ee5c9e600dbd |
28-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r6161 "Drop the DataChannel message if it's received when the channel is not open."
The spec does not say the DataChannel has to be open to receive a message.
TBR=pthatcher@google.com
BUG=crbug/363005
Review URL: https://webrtc-codereview.appspot.com/16569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6264 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
|
| f666ecc60d48a3c037b77a0c6e6d40b46567aa76 |
27-May-2014 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling flaky libjingle tests after fixit week.
BUG=webrtc:3316,webrtc:3317,webrtc:3318
TBR=fischman@google.com
Review URL: https://webrtc-codereview.appspot.com/12569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6250 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/sharedexclusivelock_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
|
| 727ff698298677ea221a176698cb47e8648da621 |
24-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67872893-> 67873348
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/call/console.cc
|
| 75cb3dc5f2a104261e5a86e3cb1cf1a42cf355c0 |
24-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67869540-> 67872893
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/call/console.cc
xamples/call/console.h
|
| b445f26f24cbfc24a6bf9a18122d778417abfb75 |
24-May-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6.
BUG=N/A
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21499007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6242 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 39eccefbde3f096eec57efb4ee5dbcce6528fba5 |
23-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable ChannelManagerTest.StartupShutdownOnUnstartedThread
The test is testing a scenario that shouldn't happen.
BUG=3388
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6238 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channelmanager_unittest.cc
|
| 7aa1a4767f14b58924822a0b7b30b265870fa806 |
23-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67848628-> 67848776
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6237 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
2p/base/constants.cc
2p/base/constants.h
2p/base/transport.cc
|
| e5063b173303e9ee6c2246d2aa42a1480902b867 |
23-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Thread: delete racy API (Release()) and fix racy code (started()).
- Thread::Release() wrote a local variable on the calling thread but read it on
another thread, with no synchronization. Happily it has no non-test callers
so deleting it instead of trying to fix it (see bug for details).
- Thread::started_ similarly was racily being written to; replaced with a
running_ Event, and hid the accessor except for tests & legacy callers,
with a note about why it's a bad idea.
webrtc/base patched with:
git diff origin --relative=talk/base | patch -p1 -dwebrtc/base
followed by manual merge of 3 thunks that ran afoul of naming differences
between talk/base and webrtc/base.
BUG=3388
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6236 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/signalthread_unittest.cc
ase/thread.cc
ase/thread.h
ase/thread_unittest.cc
ession/media/channelmanager.cc
|
| 18f41b8eb4da76d6ab4b8c6bf142412dc4a4f4f4 |
23-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
Example files that this makes ok instead of flagging include:
talk/base/signalthread_unittest.cc
talk/base/thread_unittest.cc
webrtc/base/signalthread_unittest.cc
webrtc/base/thread.cc
webrtc/base/thread.h
webrtc/base/thread_unittest.cc
BUG=1027
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6235 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| 706152dcc9fc56c8bb05465487717bd5a84badb2 |
23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix uninitialized reads in IsDefaultBrowserFirefox
BUG=
TEST=Local DrMemory.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19529006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6232 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/proxydetect.cc
|
| 8e755c1ad2adfd12444e2cb72b080896ae1b783d |
22-May-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed
when TURN ports are using shared socket with UDP port.
This is required as AllocationSequence maintains a map of turn ports. If the
ports are destroyed without the knowledge of AllocationSequence, sequence will
try to deliver packets to the destoyed ports.
R=jiayl@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=368877
Review URL: https://webrtc-codereview.appspot.com/14569007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6219 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
|
| f9f1bfbdaecc68594bc9ca8e52f2733e2742d3f8 |
21-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67686255-> 67689476
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/timeutils.cc
ase/timeutils.h
ase/timeutils_unittest.cc
edia/webrtc/webrtcvideoengine.cc
|
| ce4201df52a3f9378834e12b49b701f57e0b82c5 |
21-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67643194-> 67686255
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 000658a138d4505b9c8cd851807e959161489fe3 |
21-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot.
BUG=N/A
TBR=mcasas@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21519006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 3b7e282caad490c1eca8af3b9f61efa31d135e3c |
21-May-2014 |
mcasas@webrtc.org <mcasas@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling systematically failing
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga
TBR= pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 49a6a27bf02c07e0d1a988e93cffcb5f6705dd96 |
21-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67555838-> 67643194
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor.h
|
| 1732a591e74e1f35e19b3b1783a9fb925ed93913 |
20-May-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a UIView for rendering a video track.
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.
R=fischman@webrtc.org
BUG=3188
Review URL: https://webrtc-codereview.appspot.com/12489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCEAGLVideoRenderer.mm
pp/webrtc/objc/RTCEAGLVideoView+Internal.h
pp/webrtc/objc/RTCEAGLVideoView.m
pp/webrtc/objc/RTCI420Frame+Internal.h
pp/webrtc/objc/RTCI420Frame.mm
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCVideoRenderer+Internal.h
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/public/RTCEAGLVideoRenderer.h
pp/webrtc/objc/public/RTCEAGLVideoView.h
pp/webrtc/objc/public/RTCI420Frame.h
pp/webrtc/objc/public/RTCMediaStreamTrack.h
pp/webrtc/objc/public/RTCVideoRenderer.h
pp/webrtc/objc/public/RTCVideoRendererDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCVideoView.h
xamples/ios/AppRTCDemo/APPRTCVideoView.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/Info.plist
ibjingle.gyp
ibjingle_examples.gyp
|
| 40bc7779aa7caa0ecac413d768b89a9315fb87f1 |
19-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk_base: remove lock inversion between MessageQueue and MessageQueueManager.
Removes the concept of a MessageQueue being "active" in favor of considering all
live MQ's to be active.
(previously a MQ was active starting from the first Post to it and stopped being
active in its dtor).
BUG=3230
R=sriniv@google.com
Review URL: https://webrtc-codereview.appspot.com/21489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6190 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/messagequeue.cc
ase/messagequeue.h
ase/thread.cc
|
| cb711f77d2ff9ebd42678869a73353809b3af66e |
19-May-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add interface to propagate audio capture timestamp to the renderer.
BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
|
| 1e019d10b8bcd96e8cf6b3d3df2730449fbed939 |
16-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix delivery error-checking missed in r6151.
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.
BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 6bfd6196ff1eac56a7f3f0191d91e06f6f9ce579 |
15-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67052073-> 67134648
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6174 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| bb6201ae4bc476d34dd48df193603d93ced176b0 |
15-May-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
TCP remote socket address should have both server hostname and IP address.
Hostname is necessary when we are creating TLS based socket, for certificate
verification.
BUG=https://code.google.com/p/chromium/issues/detail?id=306285
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6165 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/turnport.cc
|
| a150bc9bbf28a4fdb7171746d0a60d550a9bb06a |
15-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine.
Enables applications that don't want to pay the init/startup cost or request
extra permissions (e.g. audio-only app, or DataChannel-only app).
BUG=3234
Review URL: https://webrtc-codereview.appspot.com/15489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| ef5a752c29f413ee1b90269263d4cae6ff693ac8 |
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67043374-> 67044055
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6163 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/gunit.h
ase/testutils.h
ase/unittest_main.cc
edia/base/testutils.cc
|
| 3e924683d424f82b22ff1b61edaa560ac2675112 |
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67037200-> 67043374
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6162 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| 4f5801494d7ea3e8ef0df545404cb45a4a0558b6 |
14-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Drop the DataChannel message if it's received when the channel is not open.
It may happen when the JS has closed the channel on the signaling thread while messages are received on the worker thread and posted before the state change is pushed to the worker thread.
BUG=crbug/363005
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19469005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6161 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
|
| 372701a8728cad7cffbd59403eb21d76352c1151 |
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67023528-> 67036361
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6160 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/gunit.h
ase/unittest_main.cc
|
| 688ed699e0a95e91777a15f5b507139af627f11b |
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67017551-> 67023528
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6158 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 2c98af7935cdde77c0ded19dd4fa260a2fa4bc47 |
14-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(Java): auto-WrapCurrentThread() when creating PeerConnectionFactory.
Various pieces of talk/ assume that the current Thread is ThreadManager'd
without checking this, so unconditionally wrap the caller's thread in case it
was created by Java code unbeknownst to ThreadManager.
BUG=2947
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6154 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 4e545cc24478df6dec0f73cb8f5b9e5720fbce59 |
14-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update webrtcvideoengine2.cc to use DeliveryStatus.
talk/ changes corresponding to https://review.webrtc.org/12289005/.
BUG=3228
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6151 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine2.cc
|
| 581e2172af690941d8630895380ce67dab53b31d |
14-May-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix libjingle to provide a field_trial implementation.
This completes https://webrtc-codereview.appspot.com/14489004/ by updating libjingle rules.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6149 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| cd846dd374a83af4c6dc88dea9e14b9581f50e02 |
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66924241-> 66927231
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6134 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/portallocator_unittest.cc
|
| da510c5de60d66bf28e9b74ee0d206b7cf879297 |
14-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66923202-> 66924241
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6132 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| d8af5b51c0ce59bbb8cb6ff753ed507c90eac93a |
14-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Deallocate the result of mach_host_self() when done with it, fixing a
port leak.
The port rights obtained by mach_host_self() and mach_thread_self() need
to be deallocated with mach_port_deallocate(). They consume finite
system-wide resources. This is in contrast to mach_task_self(), which is
a macro that wraps an extern global variable, and must not be
deallocated.
http://crbug.com/105513 shows the sorts of problems that can occur when
these aren't properly deallocated.
R=fischman@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15469004
Patch from Mark Mentovai <mark@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6131 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/cpumonitor.cc
|
| c14f521b1b47f33959bf589ac2937af49db74782 |
13-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66887616-> 66900106
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6130 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector_unittest.cc
|
| 3e01e0b16cbde481241b9bcfdbbdd591cd920b99 |
13-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66867790-> 66887616
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/peerconnectiontestwrapper.cc
|
| b5a22b14648c53874b4b76368a1a2271d985e875 |
13-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r6110 and r6109.
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.
BUG=
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/codec.cc
edia/base/codec.h
edia/base/codec_unittest.cc
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvideochannelfactory.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| eaf2bd916bb2c15fe4bb4e102c6fdb28c7bd6e8f |
13-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66813165-> 66836233
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6113 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| d37bcfa8820507e7f93052a4cac62acdd80978e9 |
13-May-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Changed enums to less generic names.
IPv4/IPv6 will be sent when RegisterUMAObserver is called. This is done
as Initialize is not called through interface.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14469006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6112 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/umametrics.h
|
| 17911dca8099707b5c050741a108b95b79a4da66 |
12-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66798415-> 66813165
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/codec.cc
edia/base/codec.h
edia/base/codec_unittest.cc
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvideochannelfactory.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 0df2ea064fad3b43a4cdca24faf982fd8078b322 |
12-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rollback of r6108
BUG=N/A
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6109 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/codec.cc
|
| a7f70a487f715b10f11918845553a99560b7a9c2 |
12-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initialize bitrates in ValidateCodecFormat.
Attempt to un-break a Visual Studio build (unknown version) that
incorrectly reports that these are potentially uninitialized.
BUG=
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15469005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6108 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/codec.cc
|
| d266a2020f9e86a787eada77d458ee75426d68af |
12-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial wiring of new webrtc API in libjingle.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/codec.cc
edia/base/codec.h
edia/base/codec_unittest.cc
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvideochannelfactory.h
edia/webrtc/webrtcvideoengine2.cc
edia/webrtc/webrtcvideoengine2.h
edia/webrtc/webrtcvideoengine2_unittest.cc
|
| 0f2a22b3fa86222f894a67d1d0e08912323589fa |
09-May-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed sending metrics from PeerConnection about IPv4 and IPv6.
Reasons: 1: There is memcheck failure.
2: DoInitialize is called before RegisterUMAObserver,
which means this will be never triggered in real cases.
BUG=3326
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6097 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
|
| 8a54844333a889e555ad4283ae47a607770a073c |
09-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66624678-> 66643715
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6095 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/portallocator_unittest.cc
|
| 1cd14a4502467c5b194c9810aed6341056500f8d |
09-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66556498-> 66624678
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6093 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/portallocator_unittest.cc
|
| ca27236272bf28eb024db24b4487ba85cdb23f3c |
09-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66541346-> 66556498
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6088 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/call.cc
ession/media/call.h
ession/media/currentspeakermonitor.cc
ession/media/currentspeakermonitor.h
ession/media/currentspeakermonitor_unittest.cc
|
| 1567b8cf8cc486067c5ddf327dd3516bc8dc93e7 |
08-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66540208-> 66541346
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6085 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
pp/webrtc/umametrics.h
|
| 073dfdd10a10d9cd6415a7bec14f472a1879457f |
08-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66539128-> 66540208
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6084 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| d1ae89fae1455c59635c51361537db261184247b |
08-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66524760-> 66539128
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6083 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| ff6a3d920aa6fd5611d2a3f55d219b0dba904eac |
08-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66523887-> 66524760
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6080 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
|
| f7026cd7c84b7c10894972ba03d8d7b9c04a99f0 |
08-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check SCTP_EWOULDBLOCK instead of EWOULDBLOCK in SctpDataMediaChannel.
usrsctp.h redefines EWOULDBLOCK to WSAEWOULDBLOCK on Windows, but usrsctp_sendv still returns the BSD EWOULDBLOCK (i.e. SCTP_EWOURLBLOCK) when sending data fails due to congestion.
We will need to revert this change when usersctp is fixed.
BUG=2866
R=juberti@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6079 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
|
| c5bb22395cc7a22b451fff0d968e7af7f759cde8 |
08-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66424806-> 66523513
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6078 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/client/basicportallocator.cc
|
| 2219037e5ebd591017fc16f6bf24d69e588e60b9 |
07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66406192-> 66424806
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6075 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| dd4742a9efbd47262e3c13f0ad805c02c921aa95 |
07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66388864-> 66406192
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6072 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| ed97bb0eb4e8b8b6a7c750d8bf5f8ad8fb5d0733 |
07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66340694-> 66388864
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
edia/base/constants.cc
edia/base/constants.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| f9277a9381815ecfe8368a45aa891eb8edf63503 |
07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66326258-> 66340694
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6069 4adac7df-926f-26a2-2b94-8c16560cd09d
mpp/constants.cc
mpp/constants.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient_unittest.cc
|
| 861d4b0de99f6d67c0f672ce94c1714cc7236bd8 |
07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66322380-> 66326258
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6067 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/devicemanager.cc
|
| 0581f0ba0a28fb4a85019efda2dd3fadcd081172 |
06-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66303009-> 66322380
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6065 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
|
| a18b4c96afef956f5b570f671d92624911f17f77 |
06-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66301332-> 66303009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6064 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| e65c9a6e67e589e27d08e8541603db1ef898976a |
06-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66299810-> 66301332
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6063 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/devicemanager.cc
|
| 0b53bd29af83eebc4c5a8f5c43f9bf0ae49d898d |
06-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66294299-> 66299810
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6062 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/webrtc/webrtcvideoengine.cc
|
| 150835ea34e1ee42d7af993fdcb82d98ff110d78 |
06-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66236292-> 66294299
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 5ee0f05d5fbb3fbe4862a76ab75d08ae846e6141 |
05-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66138442-> 66236292
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
ession/media/bundlefilter.cc
ession/media/bundlefilter.h
ession/media/bundlefilter_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/ssrcmuxfilter.cc
ession/media/ssrcmuxfilter.h
ession/media/ssrcmuxfilter_unittest.cc
|
| 41451d4e55e9cc00c342d0ad64dcf891cfb24622 |
03-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66106643-> 66138442
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6049 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
|
| cc06c75f284416c39f8118b75a3ee96fbf6344c0 |
02-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66100938-> 66106643
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6046 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/fakenetwork.h
ase/network.cc
ase/network.h
ase/network_unittest.cc
|
| 13d6776c46642e708b9a7e8e72c7457b8316d5e2 |
02-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66098243-> 66100938
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6045 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine.cc
|
| 0d34f1446a93f964cf6e221ca0ebd63935950b14 |
02-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66033941-> 66098243
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 14ea7e8922e2be5f51fdaa2494a64a0f39771860 |
01-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation.
- tap display to toggle visibility
- increased getStats frequency to 1hz.
R=glaznev@google.com
Review URL: https://webrtc-codereview.appspot.com/19419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| dd92feb6ddee3be376ce7566ccebd53feb0d6152 |
01-May-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): send the created SDP, not the local description after setting it
This is required to allow explicit filtering of ICE candidates.
BUG=3288
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 9c16c39e613ebc5cdfa8ca5818a62ef5c3b18bd7 |
01-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Sets the SCTP port codec in the native SessionDescription.
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.
BUG=3141
R=juberti@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
edia/base/codec.h
edia/base/codec_unittest.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
ession/media/mediasession.cc
ession/media/mediasession.h
|
| 53d82350c52372890a333e321e548c8b1b539ebd |
01-May-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Ignore identical remote fingerprint in DtlsTransportChannelWrapper::SetRemoteFingerprint.
Trying to set the same remote fingerprint could happen during renegotiation and should not fail.
BUG=crbug/362431
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6035 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel_unittest.cc
|
| ff2733204dd2cc894206716e111dcffabc8898f2 |
30-Apr-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement ObjC DataChannel wrapper
R=fischman@webrtc.org
BUG=3112
Review URL: https://webrtc-codereview.appspot.com/16369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannelinterface.h
pp/webrtc/objc/RTCDataChannel+Internal.h
pp/webrtc/objc/RTCDataChannel.mm
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/public/RTCDataChannel.h
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/objc/public/RTCPeerConnectionDelegate.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
ibjingle.gyp
ibjingle_tests.gyp
|
| 740e6b339a070bd571a559f6d7aee4c604fd4c5e |
30-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65843899-> 65880186
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6029 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 7c82adae6171ea0a9bf96856e0bbb67108e1e121 |
30-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16379004
Patch from Bridger Maxwell <bridgeyman@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppClient.m
|
| a86c42c42443056ca0a0a751098b0746bb86ff73 |
29-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
libjingle_unittest now compiles and passes on iOS! (reland of r5986)
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && \
~/src/ios-deploy/ios-deploy -d -u -v -b \
~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app
Note that the test's use of signals means that lldb will break in the middle
of the suite. To ignore these signals tell lldb:
pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue
BUG=3241
R=kjellander@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6025 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/iosfilesystem.mm
ase/optionsfile_unittest.cc
ase/physicalsocketserver.cc
ase/physicalsocketserver_unittest.cc
ase/socket_unittest.cc
ase/unixfilesystem.cc
ase/unixfilesystem.h
uild/ios_test.plist
uild/ios_tests.gypi
ibjingle.gyp
ibjingle_tests.gyp
|
| 681f787cc4651680c82aa3b13af49666c1b97c55 |
29-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65752960-> 65813736
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6023 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/scoped_ref_ptr.h
|
| f04a6ea7339bb570d3419cf77111b0ce60018c80 |
29-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
MediaCodecVideoEncoder: limit MediaCodec bitrate to 95% of requested to avoid overshoot.
BUG=3194
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/17379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6021 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
|
| af6640fce73fe0945b749ae8db3ddf6fc3d599a5 |
28-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65729829-> 65752960
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| f27fdeb9c906ba80a10f3638a28b73a757fcef3f |
28-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): don't initialize process-globals more than once.
BUG=3257
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 7d825e9b2c26e92e7e866c61f5e2bd6f68d7f904 |
28-Apr-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "libjingle_unittest now compiles and passes on iOS!"
This reverts commit r5986 as it fails compilation on Mac
(non-iOS). The failure was not discovered on the commitbots
since they don't clobber their builds.
BUG=3241
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5997 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/iosfilesystem.mm
ase/optionsfile_unittest.cc
ase/physicalsocketserver.cc
ase/physicalsocketserver_unittest.cc
ase/socket_unittest.cc
ase/unixfilesystem.cc
ase/unixfilesystem.h
uild/ios_test.plist
uild/ios_tests.gypi
ibjingle.gyp
ibjingle_tests.gyp
|
| a0d3067575bc3c4cbf7e56b9a7f998f79e14ae76 |
26-Apr-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use CreatePeerConnection method which accepts port_allocator.
Other method will be removed, in a different CL.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20369006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/objc/RTCPeerConnectionFactory.mm
xamples/peerconnection/client/conductor.cc
|
| 95cd1551f8aa0b86d92e0417204888264fbe10b0 |
26-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
libjingle_unittest now compiles and passes on iOS!
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && ~/src/ios-deploy/ios-deploy -d -u -v -b ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app
Note that the test's use of signals means that lldb will break in the middle of the suite. To ignore these signals tell lldb:
pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue
BUG=3241
R=noahric@google.com, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5986 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/iosfilesystem.mm
ase/optionsfile_unittest.cc
ase/physicalsocketserver.cc
ase/physicalsocketserver_unittest.cc
ase/socket_unittest.cc
ase/unixfilesystem.cc
ase/unixfilesystem.h
uild/ios_test.plist
uild/ios_tests.gypi
ibjingle.gyp
ibjingle_tests.gyp
|
| 658a94595d33bd25b683575e9dc92f33fa2a7bc6 |
26-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65619249-> 65622932
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5984 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/turnport.cc
|
| ff90ed6e96fdf28e0baa4a0d272315db22f3e01a |
25-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65561104-> 65619249
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5983 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/port_unittest.cc
2p/base/turnport.cc
|
| 2b93402e36892ec77428fb2cf10a16c03bdb7d14 |
25-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65484212-> 65561104
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5978 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 3f1aa24078b91344d6afa0122bea45bc7f6b74e8 |
24-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65469804-> 65484212
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5967 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/tcpport.cc
2p/base/turnport.cc
|
| 0d915ff603430c088e23c684b1af8c617fbcb4d9 |
23-Apr-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet.
R=juberti@webrtc.org, mallinath@webrtc.org
BUG=3244
Review URL: https://webrtc-codereview.appspot.com/12299006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel_unittest.cc
|
| 504fc89f362f461f1c37a51baa3458380a63a497 |
23-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65394435-> 65417850
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/unixfilesystem.cc
|
| 19b1be159e13a5aa2ba03bc5eda7c67e50bcfb7d |
22-Apr-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Provide GetStats method in RTCPeerConnection
BUG=3144
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCEnumConverter.h
pp/webrtc/objc/RTCEnumConverter.mm
pp/webrtc/objc/RTCPair.m
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCStatsReport+Internal.h
pp/webrtc/objc/RTCStatsReport.mm
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/objc/public/RTCStatsDelegate.h
pp/webrtc/objc/public/RTCStatsReport.h
pp/webrtc/objc/public/RTCTypes.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
ibjingle.gyp
ibjingle_tests.gyp
|
| ec3d8ecdcc8c43b0833ee2c1d1c5932b815fc34e |
21-Apr-2014 |
tkchin@webrtc.org <tkchin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix typo by renaming RTCSessionDescriptonDelegate -> RTCSessionsDescriptionDelegate
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5946 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/objc/public/RTCSessionDescriptionDelegate.h
pp/webrtc/objc/public/RTCSessionDescriptonDelegate.h
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.h
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
ibjingle.gyp
|
| 54fd70046d671d121a71a27181ea47ad12b27d48 |
19-Apr-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ASSERT in TransportChannelProxy::SetImplementation, when
proxy already set to same transport channel impl.
Since session can call SetImplementation multiple times with or without BUNDLE, there are cases when SetImplementation is called with same impl (OnRemoteCandidates/PushdownTransportDescription/SetupMux). Also variables in
cricket::TransportProxy like |connecting_| and |negotiated_| are accessed
both between worker thread and signaling threads (which calls for bigger change
on how session interacts with Transport and TransportChannelProxy). I have a created a separate bug to address later issue.
Also if single thread used as worker and signaling thread, we can end up
calling SetLocalDescription and OnRemoteCandidates in same call sequence, which
will end up calling SetImplementation twice.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5944 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/transportchannelproxy.cc
|
| 8e5ec52e76dd3dd352ef05f7498fba9d06244afe |
19-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65152644-> 65219629
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5941 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/turnport.cc
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
|
| 29540b18795a14c3eaa3bcdbd74b46239a1d2055 |
18-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "PeerConnectionFactory: delay deletion of owned threads."
This reverts r5933 because it broke
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/1598
BUG=3100
Review URL: https://webrtc-codereview.appspot.com/12159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5935 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory_unittest.cc
|
| 1a87f529a2f63bb890d1679fa44d67b42dc0a4d6 |
18-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65151416-> 65151642
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/turnport.cc
|
| cea024d6728c5bc897b542b90de5f55e75cf3fbd |
18-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionFactory: delay deletion of owned threads.
Since PeerConnection holds a ref to its creating PeerConnectionFactory, it's
possible for ~PeerConnectionFactory() to be run on its signaling thread.
Deleting a thread from within that thread is sad times, so don't do it.
It would be nicer to avoid having PeerConnection hold a ref to the factory,
and instead require the user to keep the factory alive. Unfortunately that
changes the contract on PeerConnection{,Factory} and it's unclear how to vet
existing callers for safety.
BUG=3100
R=juberti@webrtc.org, noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/11289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5933 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory_unittest.cc
|
| aeb0c28193a012f7431edd36f96937510a555fc8 |
17-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES".
BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5931 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| 0b3c6c3191d40c9efe38c8a6ab3f8642dd4e8583 |
17-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65086785-> 65104022
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 39b868bad3e895bcd0a2df2d6285d9f81c0eb302 |
17-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 65055925-> 65086785
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/turnport.cc
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
|
| 8f88f20af239805155f1540d7da53e106dd195d7 |
16-Apr-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Expand the test max wait time from 1000ms to 2000ms.
The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck.
BUG=2838
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 36eda7cf0e16656fb4fcb7dd5e93b5555b824e56 |
15-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329, where the m line from firefox have a space at the end.
For example:
"m=application 38233 DTLS/SCTP 5000 "
BUG=3212
TEST=manually try to use DataChannel between FF 28 and Chrome with rtccopy.com
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5915 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
|
| 1fd5b45a0eb1359fa829a0d4cf93c48d47f3e519 |
15-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64956819-> 64982143
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5910 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| 190b72a350304baaf2f19f3f82e185e98eab567b |
15-Apr-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make libjingle Android example build without sourcing envsetup.sh
See https://webrtc-codereview.appspot.com/11799004
for full details (separate to avoid webrtc+talk changes in same CL).
BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5908 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| ad4440a64ec122e16b848009c9cddfc4df98f475 |
15-Apr-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
In shared socket mode, use udp port as default receiver even if
stun server address is not set.
This can happen in a loopback scenarios where clients do not need
to provide any server information.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5906 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel_unittest.cc
2p/client/basicportallocator.cc
|
| 505f400f2765f4caf7da86f7293f8e80f0dde5a1 |
14-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64909599-> 64919255
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5905 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/iosdeviceinfo.cc
|
| e98598d3f0e35e68c84c69e904cef31e2222d907 |
14-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of
adding or renaming files. If you're doing\n# structural changes, please get a
review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >>
$d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the non-talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5904 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
uild/OWNERS
|
| 1da6047132bca9dcc1015951ce53fa6b31cc49e9 |
14-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64813990-> 64909599
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5900 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/iosdeviceinfo.cc
|
| cf0b46c762171ffbb26a7387483fcb4d55918b0e |
14-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
iosdeviceinfo.cc: remove unnecessary file
The do-nothing implementation in this file is already present in
mobiledevicemanager.cc (shared with Android) so this isn't adding value, and
causes duplicate-symbol errors under some compilers.
BUG=3201
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5899 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
edia/devices/iosdeviceinfo.cc
|
| f875f15afb5013e45b1af295b15ef4853c46a53b |
14-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64709629-> 64813990
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/relayport.cc
2p/base/stunport.cc
2p/base/stunport_unittest.cc
2p/base/tcpport.cc
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/connectivitychecker_unittest.cc
2p/client/portallocator_unittest.cc
mpp/chatroommoduleimpl.cc
mpp/pubsubtasks.cc
mpp/pubsubtasks.h
mpp/rostermoduleimpl.cc
|
| b884eb611803b4720e55bdd8b51602edf7061061 |
10-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64630087-> 64709629
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5884 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| 8dce41b3c6a5ecff583cb8d6af3c5102d66c41dd |
10-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove erronuous commit message from auto sync.
BUG=N/A
TBR=kjellander@webrtc.org
http://webrtc-codereview.appspot.com/11639004/
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5883 4adac7df-926f-26a2-2b94-8c16560cd09d
ommit_message.txt
|
| 15192f909e5a7e43287d2ec6cbb567c59afba7ce |
10-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64594651-> 64630087
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d
ommit_message.txt
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| 8f89497949fe9fd710640fa0e095beb909e2c1c9 |
09-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove erronuous commit message.
BUG=N/A
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5875 4adac7df-926f-26a2-2b94-8c16560cd09d
ommit_message.txt
|
| 61c1b8ea32d1801384151286ad8bd4eeccacf34b |
09-Apr-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64585415-> 64594651
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/webrtcsession.cc
ommit_message.txt
2p/base/dtlstransportchannel.cc
2p/base/session.cc
2p/base/session.h
|
| f824fde36f373ef031c6c606aa74383522c9807c |
09-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64326665-> 64585415
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5864 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/latebindingsymboltable.cc.def
ase/latebindingsymboltable.h.def
|
| 74a7c482b998db083ee9dccaba92758a918da52b |
07-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes unused thread causing compiler warnings.
BUG=N/A
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5859 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread_unittest.cc
|
| 4e393070be2288596170e4ac21783785ab511466 |
07-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure.
BUG=2687
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5858 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
|
| 09b0c10eed36281e2c4990abfc8953956c4d1dc6 |
05-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Talk: fixes warning: local variable is initialized but not referenced due to only using the variable in question for asserts.
BUG=N/A
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5848 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/win32regkey.cc
|
| d1fe6b728ef50ef90e386625bee138bfb361c036 |
04-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): fix a couple of SDP-related regressions.
- r5834 made it so that empty fields are a fatal SDP parsing error, exposing
opportunities for improvement in the preferISAC; changed split/join to use
\r\n instead of \n and now omitting the trailing space on the m=audio line
that triggered the new failure.
- DTLS requires a different role for each endpoint so conflicts with loopback
calling. apprtc.py suppresses DTLS for that reason in loopback calls, so the
android demo app now only enables DTLS by default if it is not suppressed by a
constraint (matching Chrome).
BUG=3164,3165,2507
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5847 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| f5bebd40f38d3d35465dc6fc1f4c8f869688b048 |
04-Apr-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64247466-> 64326665
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5845 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/genericslot.h
ase/genericslot.h.pump
ase/genericslot_unittest.cc
ase/httpserver_unittest.cc
ase/ipaddress_unittest.cc
ase/proxydetect_unittest.cc
ase/thread_unittest.cc
edia/base/videocapturer.cc
edia/base/yuvframegenerator.cc
edia/devices/macdevicemanager.cc
edia/webrtc/webrtcvideoengine.cc
2p/base/session_unittest.cc
2p/base/stun_unittest.cc
2p/base/turnserver.cc
2p/client/connectivitychecker.cc
2p/client/connectivitychecker_unittest.cc
ession/media/channel_unittest.cc
ession/media/mediasessionclient_unittest.cc
mpp/hangoutpubsubclient.cc
|
| 148149138dbc4c619230499b9a0a93b665285823 |
03-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64147530-> 64247466
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5835 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
2p/base/port.cc
|
| 5e760e7b94af0ecf3abbb793a793c2c551badece |
03-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check the return value of the FromString call and return failure when then value is invalid. I.e. uses
bool FromString(const std::string& s, T* t)
instead of
T FromString(const std::string& str)
Before this change we will silently continue the parsing and take whatever default value returned by FromString.
TEST=new tests
BUG=2507
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5834 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
|
| e387771b98bbdca223e3fa37ccb0b754d8504a4a |
03-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove webrtc_unittest.cc from talk presubmit script.
BUG=
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5833 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| 05e7b44b83f9f12a827646c496f5d6ae796b4b99 |
01-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63948945-> 64147530
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5825 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
|
| 49c5ba32bb885918cb6801d2ab47f29380bad67f |
31-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(iOS): now works in the iOS Simulator!
...which has no camera device emulation or pass-through, so no local video
view.
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCVideoRenderer.mm
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
|
| 61e78fca6cb3499fda9fce4a19d0f62ead8afbe8 |
31-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(iOS): remote-video reliability fixes
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL. Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports. Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof). Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.
Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
is always true (yay ObjC!).
- Auto-scroll messages view.
BUG=3117
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10899006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/GAEChannelClient.h
xamples/ios/AppRTCDemo/GAEChannelClient.m
xamples/ios/AppRTCDemo/en.lproj/APPRTCViewController.xib
xamples/ios/AppRTCDemo/ios_channel.html
|
| fe16488184910a7a18895d52b837b5308ad0cc49 |
28-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints.
This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640).
BUG=2774
R=jiayl@chromium.org
Review URL: https://webrtc-codereview.appspot.com/10749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 4f2bd68744583ff9ac0023f64f4600288c0cde03 |
28-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Silence pointless LS_WARNING about port 0 for active-only candidates.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5808 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
2p/base/transport.cc
|
| 987f2c9aae7f4dd3ce5eb46fbe560bc584231195 |
28-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63913264-> 63948945
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5807 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
|
| f7d501d48af4781a8a82c61851d1737def297d0a |
28-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63884381-> 63913264
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5805 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| a5586b50e5cc19c0721a5d62ba8e950d16f637f9 |
27-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Protect ENABLE_PROFILING to fix profiling=1.
Chromium defines ENABLE_PROFILING under the gyp flag profiling=1. This
corrects the resulting mulitple defintion error:
../../talk/base/profiler.h:61:9: error: 'ENABLE_PROFILING' macro redefined [-Werror]
#define ENABLE_PROFILING
and allows us to use profiling=1 in standalone builds.
TESTED=build passes with profiling=1
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5804 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/profiler.h
|
| cfe5e9c894694ea451c5559b3147389359833188 |
27-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63837929-> 63884381
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5800 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| b0ecc1c6fb107b9032611870eeae8afde3e0a5d2 |
26-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63777286-> 63837929
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5797 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/videosource.cc
pp/webrtc/videosource_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/win32toolhelp_unittest.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| b25576a75b50110ce18643c457a37eee348ac66e |
26-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk/: enable _DEBUG in Debug for all posix
Chromium's build/common.gypi defines _DEBUG for Debug builds _except_ on
(OS=="mac" OS=="ios"). But libjingle uses _DEBUG on all platforms so define it on all posix (chromium covers non-posix separately and fine).
BUG=webrtc:3101
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/10699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5795 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| 1ca08f65e358c67b7ebb8e89434cef51cf9196e5 |
26-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly.
BUG=3121
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCVideoView.h
xamples/ios/AppRTCDemo/APPRTCVideoView.m
|
| 5fb7428496d5bf6e0ef15ce15832057051f9312b |
26-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63775799-> 63776369
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5789 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
|
| a92fd74f40cac57456cc034db454b358fd3474e2 |
26-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63773382-> 63775799
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5788 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| dce3feb0b02bf1b7809f6247943979094de88593 |
26-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63738002-> 63773382
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
xamples/ios/AppRTCDemo/APPRTCVideoView.h
xamples/ios/AppRTCDemo/APPRTCVideoView.m
edia/base/mediachannel.h
edia/other/androidmediaengine.cc
edia/other/androidmediaengine.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| ae3347a546ccb356e90fc4a73b45cb8884bc3b06 |
25-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix after auto update: removed files were brought back.
BUG=N/A
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5782 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/VideoView.h
xamples/ios/AppRTCDemo/VideoView.m
|
| 76d4f389bb09609bf0b52323ebe71e6a8653f341 |
25-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(iOS): allow rooms with no incoming audio.
Also fix a compile-time warning for a leftover unimplemented method
(RTCVideoRenderer:setTransform).
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5780 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/public/RTCVideoRenderer.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
|
| 6e3dbc2a77eb96b050c4909c4206348f1b15550c |
25-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63648983-> 63738002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
xamples/ios/AppRTCDemo/VideoView.h
xamples/ios/AppRTCDemo/VideoView.m
edia/base/videoengine_unittest.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 385a722646fb73531d3e67526dc83cf1df168ede |
25-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(iOS): make ARC-clean talk/.../objc* and talk/examples/ios
- Removes a strong-reference cycle between RTCPeerConnection and
RTCPeerConnectionObserver
- Gives RTCPeerConnectionObserver a virtual dtor
- Ensures RTCPeerConnectionTest tears down correctly
- Ensures AppRTCDemo tears down correctly
This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005
BUG=3054,3055,3100
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/10499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/objctests/mac/main.mm
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.m
|
| e42b8ab1293fc5c71e741f16ae920f50fe23c301 |
25-Mar-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleanups in libjingle to make it compile with chromium_code=1
Fixed all warnings that show up when compiling libjingle
in chromium with compiling with chromium_code=1.
chromium_code=1 enables various warnings that are off by
default. Most changes are for unused variables and consts.
R=pthatcher@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsdp.cc
ase/ipaddress.cc
ase/network.cc
ase/network.h
ase/nssidentity.cc
ase/physicalsocketserver.cc
ase/physicalsocketserver.h
ase/timeutils.cc
2p/base/constants.cc
2p/base/dtlstransportchannel.cc
2p/base/pseudotcp.cc
2p/base/turnport.cc
2p/client/basicportallocator.cc
2p/client/basicportallocator.h
2p/client/httpportallocator.cc
ession/media/mediamessages.cc
mllite/xmlparser.cc
mllite/xmlparser.h
mpp/constants.cc
mpp/constants.h
mpp/xmppengineimpl.cc
mpp/xmppengineimpl.h
|
| 7fa1fcb72cc7b0d68a5e11d52724504c1cd4ac36 |
25-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10
BUG=2168
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/9709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCAudioTrack.mm
pp/webrtc/objc/RTCICECandidate.mm
pp/webrtc/objc/RTCICEServer.mm
pp/webrtc/objc/RTCMediaConstraints.mm
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCPair.m
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/RTCSessionDescription.mm
pp/webrtc/objc/RTCVideoCapturer+Internal.h
pp/webrtc/objc/RTCVideoCapturer.mm
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/RTCVideoSource.mm
pp/webrtc/objc/RTCVideoTrack.mm
pp/webrtc/objc/public/RTCMediaSource.h
pp/webrtc/objc/public/RTCVideoRenderer.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.m
pp/webrtc/objctests/mac/main.mm
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCVideoView.h
xamples/ios/AppRTCDemo/APPRTCVideoView.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/GAEChannelClient.m
xamples/ios/AppRTCDemo/VideoView.h
xamples/ios/AppRTCDemo/VideoView.m
xamples/ios/AppRTCDemo/main.m
ibjingle_examples.gyp
edia/devices/macdevicemanagermm.mm
|
| c693a2a62469148ef1bef120ebb9aa8763613765 |
24-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(iOS): fix case in #import statements.
We've been skating by on OS/X's default case-insensitive filesystem, but this
is a bit silly.
This change brought to you by:
sed -i '' 's/\+internal\.h/+Internal.h/g' $(git grep -l '+internal.h')
BUG=3088
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5764 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCAudioTrack.mm
pp/webrtc/objc/RTCICECandidate.mm
pp/webrtc/objc/RTCICEServer.mm
pp/webrtc/objc/RTCMediaConstraints.mm
pp/webrtc/objc/RTCMediaSource.mm
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/RTCSessionDescription.mm
pp/webrtc/objc/RTCVideoCapturer.mm
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/RTCVideoSource.mm
pp/webrtc/objc/RTCVideoTrack.mm
|
| 1e6cb2c5d21d778437e650170de397ace4b39b08 |
24-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63560528-> 63648983
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5762 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/nethelpers.cc
edia/base/mediachannel.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 152208adeb332321ba4c66f086fada30bf0d12a0 |
21-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63547048-> 63560528
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5753 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
|
| be7e26d22968dbd681d7fae9c217eabed3fd2459 |
21-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63503990-> 63547048
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5751 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 0c6f0f94f1444041b8759d56ee9b6e0c756d1308 |
21-Mar-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5737 "Add system wrapper dependency to libjingle targets."
Adding additional dependency is not required for libjingle targets.
> Add system wrapper dependency to libjingle targets.
> This is necessary to handle usage of STR_CASE_CMP in
> common_types.h ( as in https://webrtc-codereview.appspot.com/10099005/)
>
> TBR=wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/10309004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5744 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| 5e83c65aeeeec9a4e50f64ad3346d9d7852728b0 |
20-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63493960-> 63503990
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5743 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/physicalsocketserver.cc
|
| a8ebdb71e33571fe3b0cad4385f16bbb75b84dde |
20-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "(Auto)update libjingle 63363208-> 63493960" (r5740)
BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5741 4adac7df-926f-26a2-2b94-8c16560cd09d
eleteme.txt
|
| 5f768adc2782d6e2f1f13fe14b6448e545719de2 |
20-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63363208-> 63493960
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5740 4adac7df-926f-26a2-2b94-8c16560cd09d
eleteme.txt
|
| 979f1f8235bb393a949ca4d6956d1c17dfd5fd77 |
20-Mar-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add system wrapper dependency to libjingle targets.
This is necessary to handle usage of STR_CASE_CMP in
common_types.h ( as in https://webrtc-codereview.appspot.com/10099005/)
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5737 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_tests.gyp
|
| ffe2620c97c2b7bfe42b04453b5a981dbf1e5f06 |
19-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63352036-> 63363208
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5731 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
|
| 8b61011b6f4508237df7c825d6ba82c5dc5846f6 |
18-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63293120-> 63352036
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5720 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/externalhmac.cc
ession/media/externalhmac.h
|
| e9793ab8b872098c241a1c0bc08836e9e78607ce |
18-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63111035-> 63293120
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5717 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/videosource.cc
pp/webrtc/videosource_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
|
| 18e5911d9297848fab2de885c5425266fc8f11eb |
14-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63089643-> 63111035
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5705 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocommon.h
edia/base/videocommon_unittest.cc
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| f45a55083fbbcb3f3104368c9b611114a7fd1031 |
13-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63019975-> 63089643
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5699 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.h
|
| 827faae0ec9b51af959506479a6498f52a4c45e8 |
13-Mar-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing incorrect memset.
Found when ENABLE_EXTERNAL_AUTH is enabled in chrome.
TBR=ronghuawu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5691 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/externalhmac.cc
|
| c7bec8484bc3053073be5e845ddbf7d5c28037cd |
12-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62948689-> 63019975
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5689 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/cpumonitor_unittest.cc
edia/webrtc/webrtcvideoengine.cc
|
| 10bd88e2b52e8175f396cd7b1e6b1f5422c2cd0f |
11-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62871616-> 62948689
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5683 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
2p/base/session.cc
|
| d3d6bce9edfb708aee93518e9d5a4a222a35a935 |
10-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62865357-> 62871616
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCVideoCapturer+Internal.h
pp/webrtc/objc/RTCVideoCapturer.mm
pp/webrtc/objc/RTCVideoRenderer+Internal.h
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/public/RTCVideoRenderer.h
pp/webrtc/statscollector.cc
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.h
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/VideoView.h
xamples/ios/AppRTCDemo/VideoView.m
xamples/ios/AppRTCDemo/en.lproj/APPRTCViewController.xib
ibjingle_examples.gyp
edia/devices/devicemanager.cc
edia/webrtc/webrtcvideocapturer.cc
|
| 05376341549062f82114c96bc8d95435c00c0479 |
10-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62713454-> 62865357
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5670 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
ession/media/channel.cc
ession/media/srtpfilter.cc
ession/media/srtpfilter_unittest.cc
|
| a01daf0359e2ce113a928b6dee326b084baa4f04 |
08-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
RTCPeerConnectionTest(objc): deflake by ignoring ICECompleted.
Delivery of the state seems intermittent at best on OS/X so
ignore it until we can make it reliable.
BUG=1414,2993,chromium:348982
TBR=bemasc@chromium.org
Review URL: https://webrtc-codereview.appspot.com/9609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5664 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
|
| 13320ea3d3473dc6ca1360c9be94ffac0aab1ae5 |
07-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest(objc): expect ICE Completed state post 61460797-p10
Also a few trivial cleanups:
- No need to use STUN for a loopback test
- Reduce test call duration 10s->2s for faster iteration
- Remove obviously-irrelevant Info.plist entries (copy/pasta from iOS)
BUG=1414,2993
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/9369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5663 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objctests/Info.plist
pp/webrtc/objctests/RTCPeerConnectionTest.mm
|
| 11aab0edc2916b17f7741d2409425b46bd0fa741 |
07-Mar-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Populate VoiceReceiverInfo::delay_estimate_ms, jitter_buffer_ms, and jitter_buffer_preferred_ms to getStats.
BUG=2665
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5661 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
|
| 371243dfa3467c7be7217da4b537cc33d2bd45a6 |
07-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove std:: prefixes from C functions in talk/.
std::memcpy -> memcpy for instance. This change was motivated by a
compile report complaining that std::rand() was used instead of rand(),
probably with a stdlib.h include instead of cstdlib. Use of C functions
without the std:: prefix is a lot more common, so removing std:: to
address this.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5657 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
ase/asynctcpsocket.cc
ase/buffer.h
ase/bytebuffer.cc
ase/cryptstring.h
ase/fakesslidentity.h
ase/fileutils.cc
ase/firewallsocketserver.cc
ase/json.cc
ase/messagequeue.h
ase/natserver.cc
ase/natsocketfactory.cc
ase/nattypes.cc
ase/nethelpers.h
ase/network.cc
ase/nssidentity.cc
ase/nssidentity.h
ase/nssstreamadapter.cc
ase/nssstreamadapter.h
ase/opensslidentity.cc
ase/opensslidentity.h
ase/opensslstreamadapter.cc
ase/physicalsocketserver.cc
ase/refcount.h
ase/scoped_ptr.h
ase/sslidentity.h
ase/sslstreamadapterhelper.h
ase/stringencode.cc
ase/stringutils.h
ase/template_util.h
ase/testclient.cc
ase/transformadapter.cc
ase/versionparsing.cc
ase/virtualsocket_unittest.cc
ase/virtualsocketserver.cc
ase/virtualsocketserver.h
ase/winping.cc
xamples/call/call_main.cc
xamples/call/console.cc
xamples/call/console.h
xamples/login/login_main.cc
edia/base/filemediaengine.cc
edia/base/mediaengine.h
edia/base/rtpdump.h
edia/base/videoframe.cc
edia/devices/v4llookup.cc
2p/base/asyncstuntcpsocket.cc
2p/base/candidate.h
2p/base/pseudotcp.cc
2p/base/relayport.cc
2p/base/relayserver.cc
2p/base/relayserver_unittest.cc
2p/base/session_unittest.cc
2p/base/stun.cc
2p/base/stun_unittest.cc
2p/base/stunserver_unittest.cc
ession/media/audiomonitor.cc
ession/media/srtpfilter.cc
|
| 79047f99c1d39c6d3c16bd9bf0db3fb2eb1741bc |
07-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62691533-> 62713454
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5653 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/constants.cc
edia/base/constants.h
edia/base/fakemediaengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 2d213e450cb96f6223fd9aed20768068ba2b88f9 |
06-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62550414-> 62691533
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5652 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/externalhmac.cc
ession/media/externalhmac.h
|
| f714e7faeaa0458394898a2b2b3de8693b767ddc |
06-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove abs() use in PseudoTcp::process.
Squelches a clang 3.5 compile error for using abs() with a long instead
of labs(). Updated affected code to use uint32:s to match the sign of
m_rx_srtt.
BUG=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5651 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/pseudotcp.cc
|
| cf85f1cf3cd5ea2127cf318888a147d2afe1d985 |
05-Mar-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reorganize libjingle path variables.
BUG=chromium:343106
TEST=Trybots passing. I also successfully ran build/gyp_chromium and built Chromium with the talk/build/common.gypi modification in the checkout.
R=andrew@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5644 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
|
| b90991dade9139e5c14c3b616a9eff07b9d6fdda |
04-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62472237->62550414
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/audiotrack.h
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
edia/webrtc/webrtcvideoengine.cc
ession/media/channel.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient_unittest.cc
mpp/constants.cc
mpp/constants.h
|
| 24dae9419a3357e2a53cc0b89120eaa2bbf5ecd4 |
04-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add pthatcher@webrtc.org to talk/OWNERS.
pthatcher@ is a new member of the team with good libjingle knowledge.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5636 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| db41b4dbcdeb9a3b71b8de274db8654f3e51c99c |
03-Mar-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the deprecated GetStats method from PeerConnectionInterface.
R=fischman@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5634 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
|
| 80bbf4c3121857852bb2b11d5c2df07cf750b765 |
03-Mar-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable test SSLStreamAdapterTestDTLS.TestDTLSConnectWithSmallMtu since it does not fail anymore.
BUG=2712
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5633 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/sslstreamadapter_unittest.cc
|
| 40b3b68cdf47d7c9c3b57fca5d0a372292025f9e |
03-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62364298->62472237
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/audiotrack.h
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/fakemediastreamsignaling.h
xamples/call/call_main.cc
edia/base/mediaengine.cc
edia/base/mediaengine.h
edia/other/androidmediaengine.cc
edia/other/androidmediaengine.h
|
| 1bbfb57d71e7b02de7714c928d853994b0a2a3ea |
03-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661".
BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5631 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/OWNERS
xamples/call/call_main.cc
edia/base/mediaengine.cc
edia/base/mediaengine.h
edia/other/androidmediaengine.cc
edia/other/androidmediaengine.h
|
| 31413dc635c4448ee96dedfe78a440cc75a91166 |
03-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62364298-> 62368661
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5629 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/OWNERS
xamples/call/call_main.cc
edia/base/mediaengine.cc
edia/base/mediaengine.h
edia/other/androidmediaengine.cc
edia/other/androidmediaengine.h
|
| d3dc424fe5f330be273065fa1fee0ebca0f0771d |
01-Mar-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread.
These callbacks are called from signal thread already. There is no point
in posting messages on the same thread again.
BUG=2922
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5626 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/peerconnection.cc
pp/webrtc/webrtcsession_unittest.cc
|
| bcfc1670d6454e547a79b983162e363a3a54f1dd |
01-Mar-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): don't send local SDP until it's set.
This fixes a race condition where the remote participant could receive the
offer, create & set its answer locally, send it back, and then try to set the
answer before the local set completed. Observed intermittently in loopback
calls when setLocalDescription is intentionally delayed (debugging something
else).
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5625 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| b8395ebe1419e97d8f5f9cf28583e2fa6b3a8048 |
28-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62293974-> 62364298
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtc.scons
ase/criticalsection_unittest.cc
xamples/peerconnection/peerconnection.scons
ibjingle.scons
ain.scons
edia/base/hybridvideoengine_unittest.cc
edia/webrtc/webrtcvideoengine.cc
ite_scons/site_tools/talk_linux.py
ite_scons/site_tools/talk_noops.py
ite_scons/talk.py
|
| 806768a6ca82ed0a38ec95cc9c11531bc7d3f033 |
27-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62281784-> 62293974
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5619 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 704bf9ebec9c9425e1898f6c3f15eff685175b23 |
27-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 62063505-> 62278774
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
ase/bandwidthsmoother.cc
ase/criticalsection_unittest.cc
ase/nssstreamadapter.cc
ase/rollingaccumulator.h
ase/rollingaccumulator_unittest.cc
ibjingle.scons
ibjingle_tests.gyp
edia/base/constants.cc
edia/base/constants.h
edia/base/mediachannel.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoengine_unittest.h
edia/devices/filevideocapturer.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/session.cc
ession/media/mediasession_unittest.cc
ession/media/planarfunctions_unittest.cc
ession/media/yuvscaler_unittest.cc
|
| eaadecaf9878dce0560a77056b7b4481772df373 |
26-Feb-2014 |
braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599.
BUG=2962
TEST=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5610 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppClient.m
|
| 79a1cff65ae2fde95c04df8b818b0249a83e788a |
25-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url".
BUG=2952
TEST=Manual
TBR=braveyao
Review URL: https://webrtc-codereview.appspot.com/9099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5605 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
|
| 91cbaa477c46b44570f6f309e3ca8b39ffe27c71 |
24-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 61966318-> 62063505
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5602 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/asyncpacketsocket.h
|
| d43aa9de7a4a2b793e5ec59c86fb0b81e4052bb0 |
22-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 61901702->61966318
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5596 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/openssl.h
ase/opensslstreamadapter.cc
edia/base/constants.cc
edia/base/constants.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/webrtc/webrtcvideoengine.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/externalhmac.cc
ession/media/srtpfilter.cc
mpp/constants.cc
mpp/constants.h
|
| a7b981843f35bb6c26cf3bc95b5a00a0b9f50a93 |
21-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702).
BUG=N/A
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamhandler.cc
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncpacketsocket.h
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
ession/media/externalhmac.cc
ession/media/externalhmac.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| ef2215110c00ee1d8225b08815bfdcee918767f9 |
21-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5590 "description"
> description
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8949006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamhandler.cc
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncpacketsocket.h
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
ession/media/externalhmac.cc
ession/media/externalhmac.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 2643805a2057b92e916bcf4f71668bc80766625e |
20-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
description
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamhandler.cc
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncpacketsocket.h
edia/base/audiorenderer.h
edia/base/fakemediaengine.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
ession/media/externalhmac.cc
ession/media/externalhmac.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
ession/media/srtpfilter_unittest.cc
|
| 571df2dca9357620e69690c562370680ddb67b6f |
20-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 61759961->61834300
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5580 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
ase/windowpickerfactory.h
|
| 5cf3e8f0f0706784b61d8d202a71b53c5d614413 |
18-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5572 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/event.cc
mpp/constants.cc
mpp/constants.h
|
| 358e3367a39a62679eba81f57171850c75b80607 |
18-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): enable HW encoder on N5 for standalone build.
Now that bug 2899 is fixed (r5562) packet-loss is recoverable. Yay.
BUG=2575
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/8869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5568 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| c2d75e07082cb6e14ba1078875f4a5a6e4a9560c |
18-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): account for thread shutdown vagaries.
Android's JVM requires threads to detach before they exit, but ONLY if
they needed to AttachCurrentThread. Conversly, threads that were
attached by the JVM (e.g. the result of making a native call from Java)
must NOT be detached by the application. This is bug 2441.
The fix for the above is to only pthread_setspecific() for threads that
Attach(), not for already-attached threads. To ensure that we only
detach Attached threads, added a GetEnv() call to ThreadDestructor(),
which revealed that Oracle's JVM can overly-eagerly clear TLS accounting
data, effectively detaching threads without their consent at shutdown.
Work around this with a specific check.
To guard against (some) regression, added a variant of PeerConnectionTest
that runs on a non-main thread. This revealed a bug in LinuxDeviceManager
which implicitly assumes its talk_base::Thread has already been
initialized. Fixed that here too.
BUG=2441
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5567 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
edia/devices/linuxdevicemanager.cc
|
| 92fdfebeddc5a1152d6c089df56a8ae4e9d9207c |
17-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 61699344.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5560 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/asyncpacketsocket.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| b8c254abd6fa784294277e2baa8298c3352faf78 |
15-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 61549749-> 61608469
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5555 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/asyncinvoker-inl.h
ase/asyncinvoker.cc
ase/asyncinvoker.h
ase/messagehandler.h
ase/thread_unittest.cc
edia/base/mediachannel.h
edia/webrtc/webrtcvoiceengine.cc
|
| c5d506a1068f48651685f6ffa835269aa461255c |
14-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): clarified README on how to launch app using adb.
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/README
|
| a3708ecdfe3c3bcd3281cd9fff70e11b6b5dce24 |
14-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest(java): unbreak following 61460797-p10
BUG=1414
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5550 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
|
| 385857dfd414dcc1fb4941218b52417808349030 |
14-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 61549749.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncpacketsocket.h
ase/asynctcpsocket.cc
ase/asynctcpsocket.h
ase/asyncudpsocket.cc
ase/asyncudpsocket.h
ase/natserver.cc
ase/testclient.cc
ase/testechoserver.h
ase/virtualsocket_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayserver.cc
2p/base/session.cc
2p/base/session.h
2p/base/session_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunserver.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
ession/media/channel.cc
ession/tunnel/pseudotcpchannel.cc
|
| b9a088b920d1ba16e0593698d4a613bb7bb5481f |
14-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 61538839.
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/8669005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectionproxy.h
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtc.scons
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/asyncpacketsocket.h
ase/fakenetwork.h
ase/fakesslidentity.h
ase/maccocoasocketserver.h
ase/maccocoasocketserver.mm
ase/network.cc
ase/network.h
ase/network_unittest.cc
ase/openssl.h
ase/openssladapter.cc
ase/openssldigest.cc
ase/opensslidentity.cc
ase/opensslstreamadapter.cc
ase/physicalsocketserver.cc
ase/socket.h
ase/thread_unittest.cc
ibjingle.gyp
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/mediachannel.h
edia/base/videoadapter.cc
edia/base/videoengine_unittest.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
2p/base/candidate.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/client/portallocator_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 0de29504ab7ac923401c8e4e154f3b72038dbcc2 |
13-Feb-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5545 "Update libjingle to 61514460"
> Update libjingle to 61514460
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8649004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtc.scons
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/asyncpacketsocket.h
ase/fakenetwork.h
ase/fakesslidentity.h
ase/network.cc
ase/network.h
ase/network_unittest.cc
ase/openssl.h
ase/openssladapter.cc
ase/openssldigest.cc
ase/opensslidentity.cc
ase/opensslstreamadapter.cc
ase/physicalsocketserver.cc
ase/socket.h
ase/thread_unittest.cc
ibjingle.gyp
edia/base/videoadapter.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine.cc
2p/base/candidate.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/client/portallocator_unittest.cc
|
| e749c9ebdb2eb2a519c72c827e70107cbc56d270 |
13-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 61514460
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/remoteaudiosource.cc
pp/webrtc/remoteaudiosource.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/webrtc.scons
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/asyncpacketsocket.h
ase/fakenetwork.h
ase/fakesslidentity.h
ase/network.cc
ase/network.h
ase/network_unittest.cc
ase/openssl.h
ase/openssladapter.cc
ase/openssldigest.cc
ase/opensslidentity.cc
ase/opensslstreamadapter.cc
ase/physicalsocketserver.cc
ase/socket.h
ase/thread_unittest.cc
ibjingle.gyp
edia/base/videoadapter.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine.cc
2p/base/candidate.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/client/portallocator_unittest.cc
|
| 3eda643a9133ef2f7768533ea96b7e3f6a34711d |
13-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack.
BUG=2912
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 540acde5b3651caee742124b28b3e8858d76a759 |
13-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): use MediaCodec for HW-accelerated video encode where available.
Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved.
Also (because I needed them during development):
- make AppRTCDemo "debuggable" for extra JNI checks
- honor audio constraints served by apprtc.appspot.com
- don't "restart" video when it hasn't been stopped (affects running with the
screen off)
BUG=2575
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/8269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
xamples/android/AndroidManifest.xml
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
ibjingle.gyp
|
| 14d80793a891a088d5221b78f07c950d0adb1d90 |
12-Feb-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionClient needs to initialize SSL.
BUG=2911
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/peerconnection/client/linux/main.cc
xamples/peerconnection/client/main.cc
|
| dd82fa726cd69cee004cb18071a5355bd8b42e5e |
11-Feb-2014 |
wjia@webrtc.org <wjia@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5516 "Thread annotation of talk_base::CriticalSection."
r5516 failed compilation on builds with enable_webrtc=0.
> Thread annotation of talk_base::CriticalSection.
>
> Also enabling -Wthread-safety in talk/build/common.gypi for clang on
> Linux. Thread annotations are compile-time checks that for instance
> certain locks are held before accessing a value.
>
> BUG=
> TEST=Local GUARDED_BY() annotations.
> R=andresp@webrtc.org, fischman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8189004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5523 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/criticalsection.h
ase/sharedexclusivelock.h
ase/signalthread.h
uild/common.gypi
ession/media/mediamonitor.h
|
| 82387e4608ade44546e4a64b61d40de079aa6ed0 |
10-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add ability to receive calls for iOS
BUG=2701
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7989005
Patch from Sajid Hussain <shussain@temasys.com.sg>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
|
| 0a7085ffc21694889b7d6efe20db18b246a0039d |
10-Feb-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Thread annotation of talk_base::CriticalSection.
Also enabling -Wthread-safety in talk/build/common.gypi for clang on
Linux. Thread annotations are compile-time checks that for instance
certain locks are held before accessing a value.
BUG=
TEST=Local GUARDED_BY() annotations.
R=andresp@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5516 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/criticalsection.h
ase/sharedexclusivelock.h
ase/signalthread.h
uild/common.gypi
ession/media/mediamonitor.h
|
| 4723dc88b3ac1743bd9a6498af414c9d4925cf25 |
09-Feb-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5511 "Revert 5510 "Disable failing libjingle_p2p_unittest..."
So, the test apparently failed right away at
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/1224/steps/libjingle_p2p_unittest/logs/stdio
> Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux"
>
> According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2
> r5505 was committed to resolve exactly these flakes.
> Let's revert the disabling and see.
>
> BUG=2907
> TBR=mallinath@webrtc.org
>
> > Disable failing libjingle_p2p_unittest test on Linux
> >
> > I realize this diables 84 test cases and for all platforms, which
> > I'm not really comfortable with. I tried finding a better way but
> > couldn't without doing significant changes to the file.
> > I think the tests either needs to be fixed or otherwise refactored
> > in order to make more fine-grained disabling possible.
> >
> > Another (too) large disabling was done by holmer@ in
> > https://webrtc-codereview.appspot.com/2227004 where he should only have
> > disabled them on Windows, if the failures in webrtc:2383 was all that
> > caused those flakes.
> >
> > BUG=2907
> > TEST=Verified this ran 0 tests:
> > out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
> > TBR=wu@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/8309004
>
> TBR=kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8329004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5513 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel_unittest.cc
|
| 607c805b8760e3c526c225acb8ded9c3bc91cd72 |
09-Feb-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll chromium_revision 245382:249215
The find_depot_tools.py is needed to workaround the import
error we get from gyp_chromium when importing it in
webrtc/build/gyp_webrtc (to avoid code duplication).
gyp_chromium introduced a dependency on it in
http://crrev.com/245412 but as we cannot sync all of Chrome's
src/tools (it's quite big), we'll work around this by
adding an empty find_depot_tools module.
The removal of the Cygwin relates to
http://crrev.com/248802 which is a step on the way to remove
Cygwin in Chromium. We seem to already be able to remove it
entirely for WebRTC though.
Changes in the isolate framework required us to update our
copies of the isolate.gypi files.
BUG=none
TEST=trybots passing on all platforms
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5512 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
|
| ce2b44532ec2fe49e2dbb2aa5106e09ad6d6bd03 |
09-Feb-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux"
According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2
r5505 was committed to resolve exactly these flakes.
Let's revert the disabling and see.
BUG=2907
TBR=mallinath@webrtc.org
> Disable failing libjingle_p2p_unittest test on Linux
>
> I realize this diables 84 test cases and for all platforms, which
> I'm not really comfortable with. I tried finding a better way but
> couldn't without doing significant changes to the file.
> I think the tests either needs to be fixed or otherwise refactored
> in order to make more fine-grained disabling possible.
>
> Another (too) large disabling was done by holmer@ in
> https://webrtc-codereview.appspot.com/2227004 where he should only have
> disabled them on Windows, if the failures in webrtc:2383 was all that
> caused those flakes.
>
> BUG=2907
> TEST=Verified this ran 0 tests:
> out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
> TBR=wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8309004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5511 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel_unittest.cc
|
| 8d2ddd00f113ac38437ee71b88f7a09ee278bfc0 |
08-Feb-2014 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable failing libjingle_p2p_unittest test on Linux
I realize this diables 84 test cases and for all platforms, which
I'm not really comfortable with. I tried finding a better way but
couldn't without doing significant changes to the file.
I think the tests either needs to be fixed or otherwise refactored
in order to make more fine-grained disabling possible.
Another (too) large disabling was done by holmer@ in
https://webrtc-codereview.appspot.com/2227004 where he should only have
disabled them on Windows, if the failures in webrtc:2383 was all that
caused those flakes.
BUG=2907
TEST=Verified this ran 0 tests:
out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5510 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel_unittest.cc
|
| cc685acbdf79877abe7d52eb02ba36903647880d |
08-Feb-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable AsyncInvokeTest.CancelInvoker test
Test is flaky.
BUG=b/12944358
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5508 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread_unittest.cc
|
| 017881065990b83be71fb54c92bf2a22428614ef |
08-Feb-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Don't use LOG() in callback.h
Because chromium is compiled with a different version of logging macros
defined in logging.h that header cannot be used in headers that can
also included from chromium code. Removed LOG_F(LS_WARNING) from
callback.h . That issue would block this code from being rolled in
chromium.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5507 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/callback.h
ase/callback.h.pump
|
| 5a59ccbb6df83176229ffaaa4128110a784e7a36 |
08-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switching to NSS random number generator and adding init method to unittests.
R=jiayl@webrtc.org, sergeuy@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5505 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/relayport_unittest.cc
2p/base/relayserver_unittest.cc
2p/base/stunrequest_unittest.cc
2p/client/portallocator_unittest.cc
|
| 9cf037b83184374230c6825e4aa407cdafaba434 |
07-Feb-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 61168196
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastreamhandler.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
ase/asyncinvoker-inl.h
ase/asyncinvoker.cc
ase/asyncinvoker.h
ase/bind.h
ase/bind.h.pump
ase/bind_unittest.cc
ase/callback.h
ase/callback.h.pump
ase/callback_unittest.cc
ase/messagehandler.h
ase/scopedptrcollection.h
ase/scopedptrcollection_unittest.cc
ase/thread.h
ase/thread_unittest.cc
ibjingle.gyp
ibjingle.scons
ibjingle_tests.gyp
edia/base/mediachannel.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoengine_unittest.h
edia/devices/v4llookup.cc
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
2p/base/stunport.cc
ession/media/channel.cc
ession/media/channel.h
|
| ea1c5ad58f0b9fb98e66df6c62a020f541ad66f5 |
06-Feb-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix gunit compilation on VS2012.
In VS2012 compiling gunit or its dependencies triggers a lot of
"'std::tuple' : too many template arguments" warnings. The workaround
for this, done for gtest already, is to define _VARIADIC_MAX=10.
BUG=2616
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5493 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 6e08228525b3f117fa551f37771951f851d64ee7 |
03-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest(java): remove the obsolete magical names of streams & tracks.
BUG=1253
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7929005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5478 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
|
| a06ebab1e19e7887d627a0da6e123d8b08fa59b6 |
03-Feb-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest(java): test SCTP DataChannels.
BUG=1408,2253,2626
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5477 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/DataChannel.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
|
| ecd622eec39c3486c7964389559a7ad6b2aa28aa |
03-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Updating libjingle.gyp after addition new files yuvframescapturer.cc.
TBR=pbos@webrc.org
Review URL: https://webrtc-codereview.appspot.com/7919006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5476 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 67ee6b9a6260fa80b83326c4b4fec8857c0e578c |
03-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60923971
Review URL: https://webrtc-codereview.appspot.com/7909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreaminterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
ase/asyncsocket.h
ase/fileutils.h
ase/linux.cc
ase/linux.h
ase/linux_unittest.cc
ase/opensslidentity.cc
ase/socket.h
ase/testutils.h
edia/base/audiorenderer.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoengine_unittest.h
edia/base/yuvframegenerator.cc
edia/base/yuvframegenerator.h
edia/devices/devicemanager.cc
edia/devices/devicemanager.h
edia/devices/yuvframescapturer.cc
edia/devices/yuvframescapturer.h
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
2p/base/dtlstransportchannel.h
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/stunport.cc
2p/base/tcpport.cc
2p/base/transport.cc
2p/base/transportchannelimpl.h
2p/base/turnport.cc
2p/base/turnport_unittest.cc
|
| 808b99b111ba15a9e212762241f0e341cee44753 |
29-Jan-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc
BUG=2749
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5460 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
edia/sctp/sctpdataengine.cc
|
| a576faf82a692c9422dcdc3278394ed25e6ee4f7 |
29-Jan-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable SCTP and use OPENSSL on Anroid and NSS on other platforms.
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.
BUG=2253
R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakedtlsidentityservice.h
ase/common.h
ase/helpers_unittest.cc
ase/nssidentity.cc
ase/nssstreamadapter.cc
ase/safe_conversions.h
ase/safe_conversions_impl.h
ase/sslidentity_unittest.cc
ase/sslstreamadapter_unittest.cc
uild/common.gypi
ibjingle.gyp
ibjingle_tests.gyp
edia/base/rtpdataengine_unittest.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
|
| 7433a088d2e97993266b66c102b0866aa90b4424 |
29-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..."
We reverted the r5421 to allow us roll webrtc to chrome without any modifications
to libjingle. Since webrtc is rolled with r5444, we can add back the original CL
and changes to libjingle will be upstreamed in the next roll.
TBR=andresp@webrtc.org
> Revert 5421 "Fix deadlock on register/unregister observer while ..."
>
> Failure to compile on Chromium Internal bots, because of API changes.
>
> http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
>
> You need to follow the steps mentioned in
> https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
>
> Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
> as mentioned in the doc.
>
> > Fix deadlock on register/unregister observer while there is a an going callback.
> >
> > BUG=2835
> > R=mallinath@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/7119005
>
> TBR=andresp@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7679004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5453 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
|
| 0dac5378e550360140b05af60608a7b1dab271dd |
28-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5447 "Update talk to 60420316."
> Update talk to 60420316.
>
> TBR=wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7719005
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5448 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface_unittest.cc
ase/asyncsocket.h
ase/fileutils.cc
ase/fileutils.h
ase/socket.h
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 752a01780914ab1da18aeb606c74f6d3b25ce3ec |
28-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60420316.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7719005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5447 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface_unittest.cc
ase/asyncsocket.h
ase/fileutils.cc
ase/fileutils.h
ase/socket.h
ibjingle.gyp
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 18586d38bcc90fa47f76e0bb54881dd889751167 |
27-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5421 "Fix deadlock on register/unregister observer while ..."
Failure to compile on Chromium Internal bots, because of API changes.
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Mac/builds/2805/steps/compile/logs/stdio
You need to follow the steps mentioned in
https://docs.google.com/a/google.com/document/d/1aHrmXECnu3-Jovc2-zYI267EaQCYz-IclYyBp9iA9Fc/edit that of a API changer.
Since I will be rolling the libjingle this week, I can push your changes along with libjingle roll, if you prepare the CLs
as mentioned in the doc.
> Fix deadlock on register/unregister observer while there is a an going callback.
>
> BUG=2835
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7119005
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5444 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
|
| 256d0ada35591c7e816de625767512d934258a0a |
24-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the check for audio codec num in WebRtcVoiceEngineTest.HasCorrectCodecs.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5430 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| ca5ff9972eab395ca14fbee8b529c2106033e7ba |
24-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-enable webrtcvoice/videoengine unittests.
TEST=try bots
BUG=
R=mallinath@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=5387
Review URL: https://webrtc-codereview.appspot.com/7149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5427 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 8d375c95b76263b7766b10fa38eb8b97c99e1682 |
24-Jan-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix deadlock on register/unregister observer while there is a an going callback.
BUG=2835
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5421 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideocapturemodule.h
edia/webrtc/webrtcvideocapturer.cc
|
| a8910d2f882730cbd0487946ce5aeda28759751c |
23-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60094938.
Review URL: https://webrtc-codereview.appspot.com/7489005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
ase/fileutils.cc
ase/fileutils.h
ibjingle.gyp
ibjingle.scons
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/mediaengine.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
2p/base/turnport_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
mpp/hangoutpubsubclient.cc
mpp/hangoutpubsubclient.h
mpp/pubsubclient.cc
mpp/pubsubclient.h
mpp/pubsubstateclient.cc
mpp/pubsubstateclient.h
|
| 0d92ef67c49d67de3c1d764dc74d3a74ba61ab5a |
22-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation".
This CL is attempting to correct those.
BUG=2810
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5411 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/public/RTCPeerConnectionDelegate.h
pp/webrtc/peerconnectioninterface.h
|
| 68cbd012160535d2a8bb6453961b7eb066902b76 |
22-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable.
BUG=2799
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5410 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface_unittest.cc
|
| 28da47c52fe8bd36f40b8557cfa9a53a2e28646f |
21-Jan-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Android example apps: fixes issue where useful failure information was suppressed.
BUG=2808
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| 2ce9a64b75929bd6c94f8f151d00cd82f41a1bc7 |
16-Jan-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples.
BUG=12545067
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/chat/Info.plist
xamples/chat/chat_main.cc
xamples/chat/chatapp.cc
xamples/chat/chatapp.h
xamples/chat/consoletask.cc
xamples/chat/consoletask.h
xamples/chat/textchatreceivetask.cc
xamples/chat/textchatreceivetask.h
xamples/chat/textchatsendtask.cc
xamples/chat/textchatsendtask.h
xamples/pcp/pcp_main.cc
xamples/plus/libjingleplus.cc
xamples/plus/libjingleplus.h
xamples/plus/presencepushtask.cc
xamples/plus/presencepushtask.h
xamples/plus/rostertask.cc
xamples/plus/rostertask.h
xamples/plus/testutil/libjingleplus_main.cc
xamples/plus/testutil/libjingleplus_test_notifier.h
xamples/plus/testutil/libjingleplus_unittest.cc
xamples/relayserver/relayserver_main.cc
xamples/stunserver/stunserver_main.cc
xamples/turnserver/turnserver_main.cc
ibjingle_examples.gyp
2p/base/relayserver_main.cc
2p/base/stunserver_main.cc
2p/base/turnserver_main.cc
|
| 4b26e2eee3e3b2a0c22946372a38f7efa6cee146 |
16-Jan-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 59676287
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/videosource_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/sslfingerprint.cc
ase/sslfingerprint.h
xamples/chat/chatapp.cc
ibjingle.gyp
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/mediachannel.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videocommon.cc
edia/base/videocommon.h
edia/base/videocommon_unittest.cc
edia/base/videoengine_unittest.h
edia/other/linphonemediaengine.h
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/dtlstransport.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/session.cc
2p/base/session.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/turnport.cc
ession/media/call.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession.h
|
| 8f19cb9fbc63e506155216f3f21619d9eed9f4b1 |
14-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5387 "Re-enable webrtcvoice/videoengine unittests."
Missed the result from the last try bot.
> Re-enable webrtcvoice/videoengine unittests.
>
> TEST=try bots
> BUG=
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/7149004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5388 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| eda682339771f436edb629c7a096918a97f73711 |
14-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-enable webrtcvoice/videoengine unittests.
TEST=try bots
BUG=
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5387 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 03cfde2d1046fd76181308a48a59e25e1a532cc6 |
14-Jan-2014 |
wjia@webrtc.org <wjia@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll Chromium 238260 -> 243863
R=andrew@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
|
| aebb1ade9d760841f243e380fa22b7ecff2d3ecc |
14-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
pRevert 5371 "Revert 5367 "Update talk to 59410372.""
> Revert 5367 "Update talk to 59410372."
>
> > Update talk to 59410372.
> >
> > R=jiayl@webrtc.org, wu@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/6929004
>
> TBR=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6999004
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncsocket.h
ase/byteorder.h
ase/logging.cc
ase/logging.h
ase/messagedigest.cc
ase/messagedigest.h
ase/messagequeue.cc
ase/socket.h
ase/sslfingerprint.h
ase/stream.cc
ase/stream.h
ase/unixfilesystem.cc
ibjingle.gyp
ibjingle.scons
ibjingle_tests.gyp
edia/base/constants.cc
edia/base/constants.h
edia/base/fakevideorenderer.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/sctp/sctputils.cc
edia/sctp/sctputils.h
edia/sctp/sctputils_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/portallocator.h
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| d7568a08c3039971cb7692147b2985a39db1cac7 |
13-Jan-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): Add OnRenegotiationNeeded support
Also:
- Make PeerConnectionObserver::OnRenegotiationNeeded() pure virtual to avoid
this sort of mistake in the future.
- Sprinkle @Override annotations on some callback definitions that were missing
them.
- Fix a JNI method-signature-lookup typo (s/(V)V/()V/) in PCOJava::OnError()
- Add an explicit ScopedLocalFrameRef to PCOJava::OnError() to match all other
C++-fired callbacks, for consistency.
BUG=2771
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/peerconnectioninterface.h
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 44461fa5cbecd556691b0ba963f95973f6abece1 |
13-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5367 "Update talk to 59410372."
> Update talk to 59410372.
>
> R=jiayl@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6929004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncsocket.h
ase/byteorder.h
ase/logging.cc
ase/logging.h
ase/messagedigest.cc
ase/messagedigest.h
ase/messagequeue.cc
ase/socket.h
ase/sslfingerprint.h
ase/stream.cc
ase/stream.h
ase/unixfilesystem.cc
ibjingle.gyp
ibjingle.scons
ibjingle_tests.gyp
edia/base/constants.cc
edia/base/constants.h
edia/base/fakevideorenderer.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/sctp/sctputils.cc
edia/sctp/sctputils.h
edia/sctp/sctputils_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/portallocator.h
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 0f3356e20b70416f13e12ef596da66f6c347eea7 |
11-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 59410372.
R=jiayl@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/sctputils.cc
pp/webrtc/sctputils.h
pp/webrtc/sctputils_unittest.cc
pp/webrtc/videosource.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncsocket.h
ase/byteorder.h
ase/logging.cc
ase/logging.h
ase/messagedigest.cc
ase/messagedigest.h
ase/messagequeue.cc
ase/socket.h
ase/sslfingerprint.h
ase/stream.cc
ase/stream.h
ase/unixfilesystem.cc
ibjingle.gyp
ibjingle.scons
ibjingle_tests.gyp
edia/base/constants.cc
edia/base/constants.h
edia/base/fakevideorenderer.h
edia/base/mediachannel.h
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/sctp/sctputils.cc
edia/sctp/sctputils.h
edia/sctp/sctputils_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/portallocator.h
2p/client/basicportallocator.cc
2p/client/portallocator_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 4625df3e3eab6634fc1521a6735f3f6c20f9b882 |
09-Jan-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix NaCl compilation
nethelpers.cc was using LOG() but didn't include logging.h
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6829005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5360 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/nethelpers.cc
|
| 4177615e87e1a92ffdf403c4ef1b09437ae4f43a |
09-Jan-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage.
Hopefully the approach of pushing/popping frames will be easier to avoid messing up than remembering to annotate every single local reference with a ScopedLocalRef.
BUG=2761
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5355 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 1794693ec8b0eeb7632b95ddfad188d16ce1b735 |
08-Jan-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): close() the throw-away DataChannel.
Otherwise, the PeerConnection remembers the channel enough to include an
m=application line in its offer SDP, causing connection to chrome to fail, since
apprtc.appspot.com doesn't include an RtpDataChannels:true constraint in its
RTCPeerConnection constructor call.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5354 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| e00265ed492c4d61f5ef04f9138739289bac6b98 |
07-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix a compile error on Android on sctpdataengine.cc.
TEST=try bots
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5350 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/sctp/sctpdataengine.cc
|
| f6d6ed0c66457170be3f3b2bc214cd7141e441a4 |
03-Jan-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 59039880.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5339 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/common.cc
ase/common.h
ase/helpers.cc
ase/ipaddress.cc
ase/nethelpers.cc
ase/network.cc
ase/nssidentity.cc
ase/nssidentity.h
ase/nssstreamadapter.cc
ase/openssladapter.cc
ase/opensslidentity.cc
ase/opensslidentity.h
ase/opensslstreamadapter.cc
ase/opensslstreamadapter.h
ase/physicalsocketserver.cc
ase/socketaddress.cc
ase/sslidentity.cc
ase/sslidentity.h
ase/sslstreamadapter.h
ase/sslstreamadapter_unittest.cc
ase/sslstreamadapterhelper.cc
ase/sslstreamadapterhelper.h
ase/stream.cc
ase/stream.h
ase/thread.cc
ase/unixfilesystem.cc
edia/base/mediaengine.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/client/basicportallocator.cc
ession/tunnel/securetunnelsessionclient.cc
|
| 000dde99c8794613e80d3b6c7252aed42d16e8c2 |
20-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Android build: make it quiet on success and not overly noisy on failure.
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
ibjingle_examples.gyp
|
| af320fd2f7d68dcf672b814e958865d5e331eeb2 |
17-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6019004
Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/public/RTCICEServer.h
|
| 5b3c67ef25c40a7866b251042e552ae54b297b75 |
16-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
objc/README: Remove outdated advice about target_os.
BUG=chromium:248168
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5979005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5302 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/README
|
| 24301a67c66e6091418e83da49cfb367ef2c6645 |
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
|
| 62451dcba06601b6643e182791bce658f79ba344 |
13-Dec-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58157731.
R=wu@webrtc.org
TBR=wu@webrc.org
Review URL: https://webrtc-codereview.appspot.com/5339005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5282 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
ase/asyncpacketsocket.h
|
| a9890800e078105f21f0a21358ee59a0b3736af6 |
13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58127566 together with
https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
ase/asyncpacketsocket.h
ase/asynctcpsocket.cc
ase/asyncudpsocket.cc
ase/natserver.cc
ase/natserver.h
ase/testclient.cc
ase/testclient.h
ase/testechoserver.h
ase/thread_unittest.cc
ase/timeutils.cc
ase/timeutils.h
ase/virtualsocket_unittest.cc
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/rawtransportchannel.cc
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/session_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/transportchannel.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/client/basicportallocator.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
|
| 2018269dc3a1c1bb01c946583ca0750ae0db68e3 |
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5274 "Update talk to 58113193 together with https://webrt..."
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
>
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
ase/asyncpacketsocket.h
ase/asynctcpsocket.cc
ase/asyncudpsocket.cc
ase/natserver.cc
ase/natserver.h
ase/sslstreamadapter_unittest.cc
ase/testclient.cc
ase/testclient.h
ase/testechoserver.h
ase/thread_unittest.cc
ase/timeutils.cc
ase/timeutils.h
ase/virtualsocket_unittest.cc
uild/isolate.gypi
xamples/android/project.properties
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_unittest.isolate
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/rawtransportchannel.cc
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/session_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/transportchannel.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/client/basicportallocator.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
|
| a129b6cd132788a931b47da3370ae473673f320d |
12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
ase/asyncpacketsocket.h
ase/asynctcpsocket.cc
ase/asyncudpsocket.cc
ase/natserver.cc
ase/natserver.h
ase/sslstreamadapter_unittest.cc
ase/testclient.cc
ase/testclient.h
ase/testechoserver.h
ase/thread_unittest.cc
ase/timeutils.cc
ase/timeutils.h
ase/virtualsocket_unittest.cc
uild/isolate.gypi
xamples/android/project.properties
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_unittest.isolate
edia/base/fakemediaengine.h
edia/base/fakenetworkinterface.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybridvideoengine.cc
edia/base/hybridvideoengine.h
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine.h
edia/base/rtpdataengine_unittest.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/rawtransportchannel.cc
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayport_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/session_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/stunserver.cc
2p/base/stunserver.h
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/transportchannel.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/base/turnserver.h
2p/client/basicportallocator.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/tunnel/pseudotcpchannel.cc
ession/tunnel/pseudotcpchannel.h
|
| df7b1d6e39fd92199e223c4554a523ba228eab5f |
11-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure.
Also silence a 'cd' that would otherwise emit the path/to/talk.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5271 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| 9ee75e9c77b467e74e470905822d0279b0e8a639 |
11-Dec-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| f41f06b916adc58745d5c5dbbd6803c7566dbedd |
11-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(java): rationalize pointer-to-jlong conversion.
In r4665 I went a bit crazy with the manual reinterpretation of a pointer to a
jlong (to avoid undefined behavior) but that's what reinterpret_cast<> is for.
So use it directly now.
Added a do-nothing DataChannel to AppRTCDemo to regression test this, since the
only repro I've found of the original bug requires ARM ABI (PeerConnectionTest
on ia32 fails to repro).
BUG=2302
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 9caf2765b285f7511d8355177c2d55209d7573e4 |
11-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58037405.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/5579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5267 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
ibjingle_tests.gyp
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/mediachannel.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videoengine_unittest.h
edia/base/videoframe_unittest.h
edia/webrtc/dummyinstantiation.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
|
| 4c3faa9d739c8b2e34182f69e862618d43a2a9f7 |
11-Dec-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable a libjingle unittest which is failing after a chromium roll out.
TBR=kjellander@google.com
BUG=
Review URL: https://webrtc-codereview.appspot.com/5559007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5264 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/sslstreamadapter_unittest.cc
|
| f9bdbe36198341d678ea22f8a47de60ee552e69a |
11-Dec-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll chromium_revision 232627:238260
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
xamples/android/project.properties
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_unittest.isolate
|
| 77507eff4fe61bd3160b669f0f5e282c320203d3 |
11-Dec-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
g++ 4.7 and later support explicit virtual overrides when building with C++11 support
enabled. However, libjingle does not detect that and makes OVERRIDE a no-op.
This CL updates base/common.h to define OVERRIDE properly when g++ 4.7 is used with
C++11 support enabled.
See this page for GCC support of C++11 features:
http://gcc.gnu.org/projects/cxx0x.html
R=fischman@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5159004
Patch from Chris Dumez <ch.dumez@samsung.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5255 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/common.h
|
| eb7def234e2fc6fd16cc627eaef813d2316c6ed6 |
09-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix compilation errors on Fedora 20.
peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.
BUG=2700
R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5019004
Patch from Victor Costan <costan@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 32f485b16a5f9c2164f18e140cfb2358e88d6700 |
05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5233 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/constants.cc
edia/base/constants.h
ession/media/mediasession_unittest.cc
|
| 57a5f64264e6b5d59062220f336bb98d2af8a578 |
05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
revert r5230
r5230 broke windows build.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5232 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/constants.cc
edia/base/constants.h
|
| a1b21cd777f7a8ec3cacefc19b6979015e1780d5 |
05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5230 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/constants.cc
edia/base/constants.h
|
| 5bc25c41fc7880545052770dbcfe67f233c9b0c0 |
05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 57692857
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
ase/macutils.cc
ase/macutils.h
ase/macutils_unittest.cc
ase/ssladapter.cc
ase/ssladapter.h
xamples/call/callclient.cc
edia/base/mediachannel.h
edia/base/rtpdataengine.cc
edia/base/streamparams.cc
edia/base/streamparams.h
edia/base/streamparams_unittest.cc
edia/base/testutils.cc
edia/base/testutils.h
edia/base/videoengine_unittest.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/turnport.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| e0034557a79bb0ba85df6b769d37a8c9ae9ff0a8 |
02-Dec-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
RTCPeerConnection(objc): avoid leaking ICE candidate on addition.
BUG=2670
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5199 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCPeerConnection.mm
|
| b43202d839818f6493abdd98dfca882373ec8220 |
22-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable PeerConnectionEndToEndTest for tsanv2 build.
BUG=1205
TEST=try
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5162 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
|
| 1977960866071e4ce406ad521ec809bcc6f8d389 |
21-Nov-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(ios): remove codesigning hack now that gyp signs by default.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5155 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_examples.gyp
|
| 364f204d16d1f10cf01b1b5543ce020c3e9961b8 |
20-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56698267.
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectionendtoend_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
pp/webrtc/test/peerconnectiontestwrapper.cc
pp/webrtc/test/peerconnectiontestwrapper.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ibjingle_tests.gyp
edia/base/videoadapter.h
edia/webrtc/fakewebrtcvoiceengine.h
2p/base/session.cc
2p/base/session.h
ession/media/call.cc
|
| 183c727bcafa7e15ce5bbd75dbbc428e599e6a6b |
13-Nov-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable datachannel_unittest.cc
the test fails to compile because it uses incorrect gmock path (as
some other tests).
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5121 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| a23f0ca4ba5105eb76b6fa30447c806812a8f3c2 |
13-Nov-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56619788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3839005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/test/testsdpstrings.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/asyncresolverinterface.h
ase/autodetectproxy.cc
ase/autodetectproxy.h
ase/autodetectproxy_unittest.cc
ase/httpclient.cc
ase/httpclient.h
ase/latebindingsymboltable.cc
ase/libdbusglibsymboltable.cc
ase/macasyncsocket.cc
ase/nethelpers.cc
ase/nethelpers.h
ase/physicalsocketserver.cc
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/client/peer_connection_client.h
ibjingle.gyp
edia/devices/libudevsymboltable.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
2p/base/basicpacketsocketfactory.cc
2p/base/basicpacketsocketfactory.h
2p/base/packetsocketfactory.h
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunport_unittest.cc
2p/base/turnport.cc
2p/base/turnport.h
2p/client/basicportallocator.cc
|
| 16d6254e8c6c865ca65cc943e03fa635dc5c6a63 |
06-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56183333.
TEST=try bots
R=sheu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/3469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5087 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videocapturer.cc
edia/base/videoframe_unittest.h
edia/webrtc/webrtcvideocapturer.cc
edia/webrtc/webrtcvideocapturer.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
|
| 7b273a545d16ce3cd26810e91751e0fff28acf71 |
04-Nov-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection iOS: update README instructions
This is needed to account for https://codereview.chromium.org/25535004/
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5079 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/README
|
| 07a6fbe83d901fc9b98579ab44e8c9632f038b36 |
04-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56092586.
R=jiayl@webrtc.org, mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/common.h
ession/media/channel.cc
ession/media/channel.h
|
| de305014c62832a382d38144a9dc518cf1d02f88 |
31-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55906045.
Review URL: https://webrtc-codereview.appspot.com/3159005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakenetworkinterface.h
edia/base/mediachannel.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/p2ptransportchannel.cc
ession/media/channel.cc
|
| f424cb8e13e3c845cc36d81e7dd17299ab98a2f7 |
30-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55863981.
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/3089006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5056 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/move.h
ase/scoped_ptr.h
edia/base/capturemanager.h
|
| cecfd1832dc375225da3f5f18ecac63006ed06bf |
30-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55821645.
TEST=try bots
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/test/fakedatachannelprovider.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/base64.cc
ase/logging.h
ase/physicalsocketserver.cc
ase/profiler.cc
ase/profiler.h
ibjingle_tests.gyp
edia/base/mediachannel.h
edia/base/streamparams.cc
edia/base/streamparams.h
edia/devices/devicemanager.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasession_unittest.cc
|
| 9ca93a8b8e811881a3c5fe31b854b873e3e5b500 |
29-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Explicitly @synthesize ObjC @properties
This is required after https://code.google.com/p/gyp/source/detail?r=1768
turned on -Wobjc-missing-property-synthesis for ninja builds (until then it
was only enabled for xcode builds) to allow chromium_deps to roll in
webrtc/DEPS.
BUG=2560
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCICECandidate.mm
pp/webrtc/objc/RTCICEServer.mm
pp/webrtc/objc/RTCPair.m
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCSessionDescription.mm
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.m
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.m
xamples/ios/AppRTCDemo/GAEChannelClient.m
|
| 850bcbe8556ef28d0276a01e56bbd382f1a81a31 |
28-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove frame_callback.h include in webrtcvie.h.
This file is about to be moved and it's not really needed. The class
I420FrameCallback is forward declared inside vie_image_process.h and
only used in talk/ for a no-op implementation that doesn't access the
pointer.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5041 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcvie.h
|
| 97077a3ab27259164eb121034b6e0ebe9ba592df |
25-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 55618622.
Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource.h
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectioninterface.h
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/atomicops.h
ase/buffer.h
ase/common.h
ase/compile_assert.h
ase/constructormagic.h
ase/cpumonitor.cc
ase/helpers.cc
ase/macutils.cc
ase/messagedigest.cc
ase/move.h
ase/natserver.cc
ase/natsocketfactory.cc
ase/nethelpers.cc
ase/network.cc
ase/network_unittest.cc
ase/physicalsocketserver.cc
ase/scoped_ptr.h
ase/socket_unittest.cc
ase/sslidentity_unittest.cc
ase/stream.cc
ase/stream.h
ase/systeminfo.cc
ase/template_util.h
ase/win32filesystem.cc
ase/win32regkey.cc
ase/win32socketserver.cc
xamples/call/callclient.cc
xamples/chat/chatapp.cc
xamples/peerconnection/client/linux/main_wnd.h
xamples/peerconnection/client/main_wnd.h
edia/base/cpuid.cc
edia/base/fakevideocapturer.h
edia/base/mediachannel.h
edia/base/videocapturer.cc
edia/base/videocommon.cc
edia/base/videocommon_unittest.cc
edia/base/videoframe_unittest.h
edia/devices/carbonvideorenderer.h
edia/devices/gdivideorenderer.cc
edia/devices/gtkvideorenderer.h
edia/devices/macdevicemanager.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctputils.cc
edia/sctp/sctputils_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvideoframe_unittest.cc
edia/webrtc/webrtcvie.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/dtlstransportchannel_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/pseudotcp.cc
2p/base/stun.cc
2p/base/stunport.cc
2p/base/testturnserver.h
2p/base/turnport.cc
2p/base/turnport_unittest.cc
2p/client/basicportallocator.cc
2p/client/fakeportallocator.h
2p/client/portallocator_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
ound/alsasoundsystem.cc
mllite/xmlbuilder.cc
mpp/xmppclient.cc
mpp/xmppengineimpl.cc
mpp/xmpplogintask.cc
|
| d371a29227710b503b450acaf8431f6369162e3f |
24-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix tsan failures for libjingle_unittest.
1) Change AsyncSocket's SignalReadEvent and SignalWriteEvent's thread mode to multi_threaded_local as they can be accessed from different threads.
2) Protect NATServer::TransEntry::whitelist.
3) Protect PhysicalSocket:error_.
Detail failures can be seen from issue 2080, comment #5.
TBR=fischman@webrtc.org
RISK=P1
TEST=try bots and tsanv2
BUG=2080
Review URL: https://webrtc-codereview.appspot.com/2669005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5026 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/asyncsocket.h
ase/natserver.cc
ase/natserver.h
ase/physicalsocketserver.cc
ase/testclient.cc
|
| 8804a29951bfeaf97a0964aa90ec69ac17820752 |
23-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread.
TEST=try bots
BUG=1205
R=henrike@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
|
| 4d7116be7ab8f0631f2e4cac1d5f56c494627056 |
22-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix tsan failures on filevideocapturer.cc.
1) init start_time_ns_ before the file_read_thread_ is started to avoid data racing as the start_time_ns_ will also be read by the file_read_thread_.
2) add CriticalSection to protect |finished_| that is accessed by FileReadThread and the main thread
Also remove the suppression for filemediaengine.cc, which has already been fixed in other cl.
TBR=henrike@webrtc.org
TEST=try bots and manual tsan v2 test
BUG=2078
Review URL: https://webrtc-codereview.appspot.com/2509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5018 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/filevideocapturer.cc
|
| 31628aae7e0d5a00e816f1f5db4b65101319a307 |
22-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Upgrade scoped_ptr to Chromium's latest version.
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 50bc5538525960d4a5346dbc6c4669e258eea28e |
21-Oct-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reenable DTLS renegotiation unittest in libjingle.
This test is failing on memcheck bots. After investigation problem per
say is not in this particular unittest and rather is in suite. So this test
is added to memcheck exclude list.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5011 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 3c5d2b43ecf80ec9619c5036938d96ca765fed52 |
18-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Thread::Stop() must be called before any subclass's destructor completes.
Update Thread documentation, fix all subclasses that had a problem.
This is to avoid a data racing between the destructor modifying the vtable, and
Thread::PreRun calling virtual method Run at the same time.
For example:
[ RUN ] FileMediaEngineTest.TestGetCapabilities
==================
WARNING: ThreadSanitizer: data race on vptr (ctor/dtor vs virtual call) (pid=2967)
Read of size 8 at 0x7d480000bd00 by thread T1:
#0 talk_base::Thread::PreRun(void*) /mnt/data/b/build/slave/Linux_Tsan_v2/build/src/out/Release/../../talk/base/thread.cc:353 (libjingle_media_unittest+0x000000234da8)
Previous write of size 8 at 0x7d480000bd00 by main thread:
#0 talk_base::Thread::~Thread() /mnt/data/b/build/slave/Linux_Tsan_v2/build/src/out/Release/../../talk/base/thread.cc:158 (libjingle_media_unittest+0x00000023478c)
#1 ~RtpSenderReceiver /mnt/data/b/build/slave/Linux_Tsan_v2/build/src/out/Release/../../talk/media/base/filemediaengine.cc:122 (libjingle_media_unittest+0x0000001b551f)
...
RISK=P2
TESTED=try bots and tsan
BUG=2078,2080
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2428004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4999 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/cpumonitor_unittest.cc
ase/dbus.cc
ase/logging_unittest.cc
ase/maccocoasocketserver_unittest.mm
ase/macsocketserver_unittest.cc
ase/signalthread.h
ase/signalthread_unittest.cc
ase/thread.cc
ase/thread.h
ase/thread_unittest.cc
ase/win32socketserver.h
edia/base/filemediaengine.cc
edia/devices/filevideocapturer.cc
edia/devices/gdivideorenderer.cc
mllite/xmlelement_unittest.cc
mpp/xmppthread.cc
|
| 1c820374942d598e810fbf7dd9501a69434dfb01 |
17-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): remove vestigial mentions of PowerManager
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2402004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4995 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/AndroidManifest.xml
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 1d1ffc9ad267d7e6e9ec9001052fd4abf29d7622 |
16-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 54898858.
TEST=try bots
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/2414004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ibjingle.gyp
ibjingle.scons
edia/base/mediachannel.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
edia/sctp/sctputils.cc
edia/sctp/sctputils.h
edia/sctp/sctputils_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/fakesession.h
ession/media/channel.cc
ession/media/channel.h
ession/media/channelmanager_unittest.cc
|
| d1cfa7149e5997010bdc8106dc2df3ff76367075 |
16-Oct-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
TSan v2 suppressions and exclusions for libjingle tests.
Add suppressions for libjingle tests so they pass under TSan v2.
Disable the following tests for TSan v2 (only) since they're failing:
* StunServerTest.TestGood
* JsepPeerConnectionP2PTestClient.*
See build logs at:
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20Tsan%20v2/
for more details.
BUG=1205,2078,2079,2080,2517
TEST=Ran a successful run of each test locally on Linux using:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release
For each test, run standing in trunk/:
TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/[libjingle_testname]
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2411004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4977 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
2p/base/stunserver_unittest.cc
|
| 6fa456f92826024921d578c7bf076e7ea2414198 |
15-Oct-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling the DTLS renegotiation test case for PeerConnection.
Currently it's failing on Linux memcheck, most likely due to timing issues.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2394006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4962 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
|
| 19f27e6a24f877fc2b0409a94b02d5f40ba3dc8c |
13-Oct-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 54527154.
TBR=wu
Review URL: https://webrtc-codereview.appspot.com/2389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/test/fakeconstraints.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ase/fakesslidentity.h
ase/nssidentity.cc
ase/nssidentity.h
ase/openssldigest.cc
ase/openssldigest.h
ase/opensslidentity.cc
ase/opensslidentity.h
ase/sslidentity.h
ase/sslidentity_unittest.cc
ase/thread.cc
uild/OWNERS
2p/base/dtlstransportchannel.cc
2p/base/transportdescription.cc
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
2p/base/transportdescriptionfactory_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 40dfbc4d3da8959bf413b4297e5da6d60182db8c |
09-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 53984350.
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/2376004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4947 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/webrtcmediaengine.h
|
| 4551b793dea4b5451cbfa13b206b6d11a25081d0 |
09-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53920541.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2371004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4945 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
ase/fakesslidentity.h
ase/nssidentity.cc
ase/nssidentity.h
ase/nssstreamadapter.cc
ase/opensslidentity.cc
ase/opensslidentity.h
ase/opensslstreamadapter.cc
ase/opensslstreamadapter.h
ase/sslfingerprint.h
ase/sslidentity.h
ase/sslstreamadapter.h
ase/sslstreamadapter_unittest.cc
ase/sslstreamadapterhelper.cc
ase/sslstreamadapterhelper.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.h
2p/base/rawtransportchannel.h
2p/base/session.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
|
| 78187525665490922748d79377bcb351579e03c0 |
08-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53856368.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2366004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4941 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
ase/latebindingsymboltable.h
ase/logging.cc
ase/logging.h
ase/md5digest_unittest.cc
ase/network_unittest.cc
ase/openssladapter.cc
ase/profiler.cc
ase/profiler.h
ase/sha1digest_unittest.cc
ase/stream.cc
ase/stream.h
ase/thread_unittest.cc
edia/base/constants.cc
edia/base/constants.h
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/hybridvideoengine.h
edia/base/mediaengine.h
edia/devices/libudevsymboltable.cc
edia/devices/libudevsymboltable.h
edia/devices/linuxdeviceinfo.cc
edia/devices/linuxdevicemanager.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
|
| 7fca2ce0979bfe5b1c60f1d6f960af9c317a5b78 |
04-Oct-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add owners to [webrtc,talk]/build and *.isolate (take 2)
After fischman@'s comments in http://review.webrtc.org/2347006/ here's another CL to clean up the redundancies and add wu@ to webrtc/build/
TEST=none
BUG=none
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2348006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4928 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
uild/OWNERS
|
| e6938185a526c3988057b0a261c7e51562f49f28 |
04-Oct-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add isolate targets for libjingle
Add .isolate file for libjingle tests and and the necessary isolate.gypi file, similar to the change in
http://review.webrtc.org/2338004/
TEST=trybots passing.
I also ran build/gyp_chromium in a Chromium checkout
with third_party/libjingle/source/talk having this patch
applied to ensure GYP processing was still working.
BUG=1916
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2353005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4926 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/isolate.gypi
ibjingle_media_unittest.isolate
ibjingle_p2p_unittest.isolate
ibjingle_peerconnection_unittest.isolate
ibjingle_sound_unittest.isolate
ibjingle_tests.gyp
ibjingle_unittest.isolate
|
| 83b9e5b32875897a66f56c26bcbebbecc71f081f |
04-Oct-2013 |
kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add owners to [webrtc,talk]/build and *.isolate
BUG=none
R=andrew@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2347006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4923 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/OWNERS
|
| 44461347574bab8ce58b39b0599f73a7765fa45f |
04-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}.
Previously it was assumed that these values were always MediaTrackConstraints but
http://dev.w3.org/2011/webrtc/editor/getusermedia.html#idl-def-MediaStreamConstraints
allows them to be boolean, too.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2352004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4918 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| a7266ca134db3b3a77ac02cb8e90e290aac90c36 |
03-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix clang build break
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2350004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4917 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 6c82e04ceed70980f2294f936a7d24bf2c06bb37 |
03-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Android standalone: remove some usages of deprecated APIs and prevent further regressions.
Also:
- Fixed WebRTCDemo UI to say "SwitchToBack" at startup since default camera is front
- Rebuild WebRTCDemo APK when resources/layout/strings change
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2337004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4916 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/project.properties
xamples/android/src/org/appspot/apprtc/AppRTCClient.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 4e65e07e4169ea8c59615817a906c3fc601a8a3b |
03-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.
BUG=1407
R=henrike@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/VideoSource.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/VideoStreamsView.java
ibjingle.gyp
|
| ddc5a19ce9cb20902def38bea91696ac14b1f61e |
03-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app.
BUG=2458
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2348004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4914 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
xamples/android/src/org/appspot/apprtc/UnhandledExceptionHandler.java
ibjingle_examples.gyp
|
| 7e4d0df8ee6fe984fbd61ea7426d3f5cd66b35e0 |
01-Oct-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection(Android): enable tracing to logcat.
BUG=1295
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2258007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4888 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/Logging.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 7e809c323a37ddd06469c6df815e4eab6c15559a |
30-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to CL 53496343.
Review URL: https://webrtc-codereview.appspot.com/2323005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4882 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
ase/natsocketfactory.cc
edia/base/rtpdataengine_unittest.cc
edia/webrtc/webrtcmediaengine.h
|
| ad81ab8861773c23407d00ce8edd02fe630dfd96 |
28-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Suppress SSL error strings on mac_asan to unbreak that build
Example borkedness:
http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingl...
Original CL for this issue is here
https://webrtc-codereview.appspot.com/2263004/
and this got reverted in here
https://code.google.com/p/webrtc/source/diff?spec=svn4874&r=4872&format=side&path=/trunk/talk/base/openssladapter.cc&old_path=/trunk/talk/base/openssladapter.cc&old=4798
Trying to land it again now.
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2318005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4875 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/openssladapter.cc
|
| a27be8e4a1f59a51ecafba71ba30ddd0bcc9f1f1 |
28-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to CL 53398036.
Review URL: https://webrtc-codereview.appspot.com/2323004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/linuxwindowpicker_unittest.cc
ase/network.cc
ase/network.h
ase/network_unittest.cc
ase/openssladapter.cc
ase/testutils.h
ase/thread.cc
ase/thread.h
ase/virtualsocket_unittest.cc
ase/windowpicker_unittest.cc
edia/base/fakemediaengine.h
edia/base/filemediaengine.h
edia/base/filemediaengine_unittest.cc
edia/base/hybridvideoengine.h
edia/base/mediaengine.h
edia/base/rtpdataengine.cc
edia/base/rtpdataengine_unittest.cc
edia/base/testutils.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/session.cc
2p/base/session.h
2p/base/transportchannelproxy.cc
ession/media/call.cc
ession/media/call.h
ession/media/channel.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
ession/media/mediasession.cc
ession/media/mediasessionclient.h
mpp/mucroomdiscoverytask.cc
|
| 4475905613cc5d05815b83d8f3cb3ff029fd3191 |
27-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable flaky RapidSpeakerChange test.
Example:
chromegw/i/internal.client.webrtc/builders/Win32%20Debug/builds/762/steps/libjingle_p2p_unittest/logs/stdio
e:\b\build\slave\win32_debug\build\src\talk\session\media\currentspeakermonitor_unittest.cc(144):
error: Value of: kSsrc2
Actual: 1002
Expected: current_speaker_
Which is: 1001
e:\b\build\slave\win32_debug\build\src\talk\session\media\currentspeakermonitor_unittest.cc(145):
error: Value of: 1
Expected: num_changes_
Which is: 2
[ FAILED ] CurrentSpeakerMonitorTest.RapidSpeakerChange (16 ms)
TBR=wu@webrtc.org
BUG=2409
Review URL: https://webrtc-codereview.appspot.com/2318004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4867 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/currentspeakermonitor_unittest.cc
|
| 2f240b43f5e6320e4c8376cd41f25cffb2b2ca38 |
25-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable some flaky libjingle base tests.
ThreadTest.Main and VirtualSocketServerTest.delay_v6
Example:
http://build.chromium.org/p/tryserver.webrtc/builders/win/builds/1234
TBR=wu@webrtc.org
BUG=2409
Review URL: https://webrtc-codereview.appspot.com/2297004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4838 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/thread_unittest.cc
ase/virtualsocket_unittest.cc
|
| f832a551ccb1c2e2edb75e6e3c5dee9f0ff00e62 |
24-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable flaky TestPartialFrameHeader.
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux%20Asan/builds/657]
TBR=wu@webrtc.org
BUG=2409
Review URL: https://webrtc-codereview.appspot.com/2286004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4832 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/devices/filevideocapturer_unittest.cc
|
| f0f92fae12376a60c416a5ac8c51da368198f184 |
24-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable flaky SendDataMultipleClocks.
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux32%20Debug/builds/719]
TBR=mallinath
BUG=2409
Review URL: https://webrtc-codereview.appspot.com/2270005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4828 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/rtpdataengine_unittest.cc
|
| 1112c30e1e5f5c7b4b517c4954ef3f15b989a996 |
23-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53057474.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2274004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4818 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/videosource_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
ase/asyncpacketsocket.h
ase/asynctcpsocket.cc
ase/asynctcpsocket.h
ase/asyncudpsocket.cc
ase/asyncudpsocket.h
ase/dscp.h
ase/natserver.cc
ase/socket.h
ase/testclient.cc
ase/testechoserver.h
ase/virtualsocket_unittest.cc
edia/base/fakenetworkinterface.h
edia/base/filemediaengine_unittest.cc
edia/base/mediachannel.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
2p/base/asyncstuntcpsocket.cc
2p/base/asyncstuntcpsocket.h
2p/base/asyncstuntcpsocket_unittest.cc
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/rawtransportchannel.cc
2p/base/rawtransportchannel.h
2p/base/relayport.cc
2p/base/relayport.h
2p/base/relayserver.cc
2p/base/session_unittest.cc
2p/base/stunport.cc
2p/base/stunport.h
2p/base/stunserver.cc
2p/base/tcpport.cc
2p/base/tcpport.h
2p/base/transportchannel.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/turnport.cc
2p/base/turnport.h
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/mediasession.cc
ession/media/mediasession.h
ession/media/mediasessionclient.cc
ession/tunnel/pseudotcpchannel.cc
|
| b533a82bf90fc02215b0a6b6b41893db57bd8878 |
23-Sep-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabled flaky tests.
BUG=2409
R=andrew@webrtc.org, mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2267005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4815 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/timeutils_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
ession/media/channel_unittest.cc
|
| d29ab4e17c4e90a5fabe7f8e0db215018f046609 |
20-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Suppress SSL error strings on mac_asan to unbreak that build
Example borkedness: http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingle_p2p_unittest/logs/stdio
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/2263004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4798 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/openssladapter.cc
|
| 76fe9309b9f532d21d889b06f722d57e0139e9d0 |
19-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use link_settings instead of all_dependent_settings to pacify xcode gyp generator
Should unbreak e.g. http://chromegw/i/chromium.webrtc.fyi/builders/Mac%20%5Blatest%20WebRTC%20trunk%5D/builds/2396
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2261004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4796 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| ccddd0a9417e5c015f0f4f6a75e1179fe33514d7 |
19-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll webrtc's chromium_revision 217707:224141
Also adds -lm for executables depending on isac since the newer clang in the
newer chromium revision requires it, and -lstdc++ for dependencies of the objc lib because newer gyp links with gcc instead of g++ for non-C++-containing libs.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2177007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4795 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 967bfff54d00f176a554bf9f955f14dde99f7bb9 |
19-Sep-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 52534915.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2251004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel_unittest.cc
pp/webrtc/localvideosource.cc
pp/webrtc/localvideosource.h
pp/webrtc/localvideosource_unittest.cc
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/remotevideocapturer.cc
pp/webrtc/remotevideocapturer.h
pp/webrtc/remotevideocapturer_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statstypes.h
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/videosource.cc
pp/webrtc/videosource.h
pp/webrtc/videosource_unittest.cc
pp/webrtc/videosourceinterface.h
pp/webrtc/videosourceproxy.h
pp/webrtc/videotrack.cc
pp/webrtc/videotrack.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsession_unittest.cc
ase/gunit_prod.h
ase/physicalsocketserver.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/devices/devicemanager.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
2p/base/sessionmanager.cc
2p/client/httpportallocator.h
ession/media/channelmanager.h
mpp/constants.cc
mpp/constants.h
mpp/hangoutpubsubclient.cc
mpp/rostermodule.h
mpp/rostermoduleimpl.cc
|
| 8d1e4d61497a47dcfe4ef5a10f17008de4690351 |
18-Sep-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Increase the dtmfsender test toleration to 100ms to avoid flaky.
BUG=2391
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/2248004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4780 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/dtmfsender_unittest.cc
|
| da79008ab4e2d1f652199ea2f927892291e28f5e |
17-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling crashing or flaky tests in peerconnection_unittest.
R=kjellander@webrtc.org
TBR=wu@webrtc.org
TESTS=trybots
BUG=2378
Review URL: https://webrtc-codereview.appspot.com/2227004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4767 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
|
| b3af8aea3e21d48049442f6982fe187ba1a6137c |
17-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Verify local and remote transport description before
negotiation.
TBR=sergeyu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2221004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4756 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/dtlstransport.h
|
| 8a1448950cc26dabe50105d7af6b37e8ca93a233 |
14-Sep-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams
BUG=2374
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2214004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4747 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/webrtcsession_unittest.cc
|
| a59696b2a5f0c138d4176249bac223ad6c4316d5 |
14-Sep-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 52300956
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2213004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ase/macsocketserver_unittest.cc
edia/sctp/sctpdataengine.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager_unittest.cc
|
| 82f014aa0bc225076516a3d77ad02deb69cfd809 |
10-Sep-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
OpenSL (not default): Enables low latency audio on Android.
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2032004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 1b476d9a5673fb1e76d9dad01882c06b94e417fe |
07-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disabling channelmanager unittest. This test is causing
TSAN error. The problem could be in thread Invoke method.
TBR=wu@webrtc.org
BUG=https://code.google.com/p/webrtc/issues/detail?id=2355
Review URL: https://webrtc-codereview.appspot.com/2190004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4700 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channelmanager_unittest.cc
|
| ab5a0912a3954abe8a22dca63e869a442be32f5d |
07-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing the build error on Windows.
Problem is in coversion from size_t to int.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4698 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/base/videoadapter.cc
|
| 1b15f4226ff417095d2146401ca71cd98ab735b3 |
07-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51960985.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2188004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4696 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/messagequeue_unittest.cc
edia/base/mediachannel.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videocommon.cc
edia/base/videocommon_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/port.cc
2p/base/transport.cc
|
| 016eec0983f903826b95176aed7f18a3ca2de89c |
06-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Unbreak build by adding new mandatory ICE username param.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2182004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4689 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objctests/RTCPeerConnectionTest.mm
|
| c31d4d03244f15681520e65bd48fd0fa5c7821a3 |
05-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(iOS): prefer ISAC as audio codec
This makes audio flow well bidirectionally to an iPod Touch (5th gen).
Also:
- Update to new turnserver JSON style:
- separate username field
- multiple URLs for the same server (e.g. both UDP & TCP)
- Added more explicit logging for ICE Connected since it's useful for debugging
- Give focus to the input field on app launch since that's the only useful
thing to have focus on, anyway.
- Fix minor typos
- Cleaned up trailing whitespace and hard tabs
BUG=2191
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2127004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4687 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/RTCICEServer.mm
pp/webrtc/objc/public/RTCICEServer.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/APPRTCViewController.m
|
| 9080518a3928285be9f94684adad064c65d2cdf3 |
05-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Restore severity precondition to logging.h.
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| ccf8b5667049a9e3165a80635071bd3bcd0a38ae |
03-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): prefer ISAC for audio codec.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2126004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4666 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 8788167b9b6ef3f8875899e59fdfd19dac5f4734 |
03-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection Java: explicitly cast DataChannel* to jlong for Java.
Otherwise on 32-bit ARM Android the nativeDataChannel param the Java ctor sees
is a 64-bit value whose low 32 bits are the pointer, and whose high 32-bits are
garbage.
BUG=2302
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2114004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4665 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| cadf9040cbb9e7bb1b73a95e43e7d228fe6b2bdb |
30-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51664136.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2148004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
pp/webrtc/localvideosource.cc
pp/webrtc/localvideosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession_unittest.cc
edia/base/fakemediaengine.h
edia/base/fakevideorenderer.h
edia/base/mediachannel.h
edia/base/testutils.h
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videocapturer.cc
edia/base/videocapturer.h
edia/base/videocapturer_unittest.cc
edia/base/videocommon.cc
edia/base/videocommon.h
edia/base/videocommon_unittest.cc
edia/base/videoengine_unittest.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/p2ptransportchannel_unittest.cc
ession/media/channel.cc
ession/media/channel.h
|
| 80b56a71e72baee88fb60cfe90cbd9b6f93f1d51 |
28-Aug-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert part of libjingle roll that caused flakiness of WebRTC tests.
BUG=crbug.com/279270
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4631 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel_unittest.cc
|
| d6fef9d3805233dd34d253036fd95fc3ed1f7113 |
27-Aug-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing SetDecodeErrorMode build error
- got introduced when reverting r4562
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2118004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4624 4adac7df-926f-26a2-2b94-8c16560cd09d
edia/webrtc/fakewebrtcvideoengine.h
|
| e3de6b1e9070806667ce9179c9607b274bf853f5 |
26-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable ObjC build by default and reenable 64-bit mac libjingle build
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2080004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| af84d782f0914a75f6f73e9c4736d940896ff132 |
26-Aug-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initialize ssl_role_ to the default role in FakeTransportChannel
constructor.
This is needed as BaseSession tests can query the transport channel
without creating dtlstransportchannel ( as they are unaware of the
underlying implementation).
This will also fix the memcheck error in webrtc bots.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2110004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4615 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/fakesession.h
|
| f1fd9d0c5ce77d706f64c3dcba5a10ee886bd5e9 |
24-Aug-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix compilation on windows after libjingle updated.
For some reason MSVC doesn't use implicit char[]->std::string
conversion when comparing char[] and std::string in EXPECT_EQ.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4611 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/p2ptransportchannel_unittest.cc
|
| 492e3154003204882d26225344ede206a16f021c |
24-Aug-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update gyp file after libjingle roll.
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2103004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4609 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| 0be6aa0665a24ec8fd5edfdddd82a707a299508c |
24-Aug-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51314459
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2100004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4608 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsessiondescriptionfactory.cc
ase/messagehandler.cc
ase/messagequeue.cc
ase/messagequeue.h
ase/messagequeue_unittest.cc
edia/base/videocapturer.cc
edia/base/videocapturer_unittest.cc
2p/base/constants.cc
2p/base/constants.h
2p/base/dtlstransport.h
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/pseudotcp.cc
2p/base/rawtransportchannel.h
2p/base/session.cc
2p/base/session_unittest.cc
2p/base/sessionmessages.cc
2p/base/stunport.cc
2p/base/stunport_unittest.cc
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.cc
2p/base/transportdescription.h
2p/base/transportdescriptionfactory.cc
2p/base/transportdescriptionfactory.h
ession/media/channel.cc
ession/media/mediasession_unittest.cc
|
| c0b1a280ab8eaebeccf5317230c3bb826454020b |
23-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some tests were not disabled correctly as it should be DISABLED_* not DISABLE_*.
TBR=wu@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/2095005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4602 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
ase/macsocketserver_unittest.cc
|
| d26f7912734fd86d03e7cd02a37add90c1756e44 |
23-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo(android): allow audio-only calls to test iOS interop
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2091004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4598 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
|
| 61b262c427d5747825d3582086786fab68d12a09 |
22-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable tests according to issues: 1205,2272,2288,2290,2291
BUG=1205,2272,2288,2290,2291
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2069005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4596 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection_unittest.cc
ase/macsocketserver_unittest.cc
|
| 7666db79fa269c6688651008edd8cf88276c0671 |
22-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51242664.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2090005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4594 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
|
| d0f4c2185b47e12df35e24c971ef12862bf9f8af |
21-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
iOS: unbreak the build following r4546
BUG=2255
R=niklas.enbom@webrtc.org, sjlee@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2078004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4577 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/ios/AppRTCDemo/APPRTCAppClient.m
|
| ebe68aad44dfd5f557f83d51d145835674781962 |
20-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix memory leak in portallocatorsessionproxy_unittest.
Remove the suppressions that have been fixed.
BUG=1972,2263
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2062005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4576 4adac7df-926f-26a2-2b94-8c16560cd09d
2p/base/portallocatorsessionproxy_unittest.cc
|
| 28ff3ee6aa34d1386f61c9277feaa41ec8c919ee |
16-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix invalid cricket::SrtpStat::FailureKey::operator<() implementation.
If operator<(a, b) returns true, then it must not be the case that
operator<(b, a) is true as well, but the old implementation would do exactly
that if a={1, 0, 0} and b={0, 0, 1}, for example.
Should fix e.g.:
[004:555] Error(unittest_main.cc:40): c:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\xtree(1746) : Assertion failed: invalid operator<
from http://chromegw/i/client.libjingle/builders/Win32%20Debug/builds/245/steps/libjingle_p2p_unittest/logs/stdio
R=juberti@webrtc.org, mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2054005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4561 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/srtpfilter.h
|
| 4d3e8b8c1b9c037a363772f30d1ffa4c6f60699c |
16-Aug-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update srtp error value in channel unittests.
TBR=ronghuawu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2053004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4557 4adac7df-926f-26a2-2b94-8c16560cd09d
ession/media/channel_unittest.cc
|
| 822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
pp/webrtc/test/fakedtlsidentityservice.h
pp/webrtc/webrtcsession_unittest.cc
edia/base/videoengine_unittest.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
|
| 97d1a988b6368a9b70f2693cb35a4ed2463b7115 |
13-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove suppressions for the cases that's already fixed.
Rename some of the suppressions to new issue.
Fix leaks in virtualsocket_unittest.
BUG=1972,1976,2100
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2010005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4536 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/virtualsocket_unittest.cc
|
| 6603736038d8555078ebbaff951cc35b80a2d491 |
13-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection::RemoveStream now removes the local stream even when it's closed. Updated the unit test accordingly.
RISK=P3
TESTED=PeerConnectionInterfaceTest.CloseAndTestMethods
TBR=fischman_webrtc
Review URL: https://webrtc-codereview.appspot.com/2018005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4535 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnectioninterface_unittest.cc
|
| 32001ef124f5082651c661965dc5d75d7f06a57b |
13-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection shutdown-time fixes
- TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted.
- PeerConnection::RemoveStream() now removes streams even if the
PeerConnection::IsClosed(). Previously such streams would never get removed.
- Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base
pointers are dispatched virtually.
- VideoTrack.dispose() delegates to super.dispose() (instead of leaking)
- PeerConnection.dispose() now removes streams before disposing of them.
- MediaStream.dispose() now removes tracks before disposing of them.
- VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API)
- AppRTCDemo.disconnectAndExit() now correctly .dispose()s its
VideoSource and PeerConnectionFactory.
- CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles
or missing .dispose() calls) in the Java API.
- Create & Return webrtc::Traces at factory birth/death to be able to assert
that _all_ threads started during the test are collected by the end.
- Name threads attached to the JVM more informatively for debugging.
- Removed a bunch of unnecessary scoped_refptr instances in
peerconnection_jni.cc whose only job was messing with refcounts.
RISK=P2
TESTED=AppRTCDemo can be ended and restarted just fine instead of crashing on camera unavailability. No more post-app-exit logcat lines. PCTest.java now asserts that all threads are collected before exit.
BUG=2183
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2005004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/MediaSource.java
pp/webrtc/java/src/org/webrtc/MediaStream.java
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/java/src/org/webrtc/VideoCapturer.java
pp/webrtc/java/src/org/webrtc/VideoTrack.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/mediastreaminterface.h
pp/webrtc/peerconnection.cc
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
2p/base/tcpport.cc
|
| a5506690b408794a122eee6d06ebebb75a2d4287 |
12-Aug-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50733053.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2017004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4532 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel_unittest.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession_unittest.cc
2p/base/dtlstransportchannel.cc
2p/base/dtlstransportchannel.h
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/p2ptransportchannel.cc
2p/base/p2ptransportchannel.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/port.cc
2p/base/port.h
2p/base/port_unittest.cc
2p/base/portinterface.h
2p/base/portproxy.cc
2p/base/portproxy.h
2p/base/rawtransportchannel.h
2p/base/session.cc
2p/base/session.h
2p/base/transport.cc
2p/base/transport.h
2p/base/transport_unittest.cc
2p/base/transportchannel.h
2p/base/transportchannelimpl.h
2p/base/transportchannelproxy.cc
2p/base/transportchannelproxy.h
2p/base/transportdescription.h
ession/media/channel.cc
|
| dd14b2add1c067c4af0ebfc89cb00030ae8ef15e |
12-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time.
- $JAVA_HOME / java_home missing or not pointing to a JDK
- Multiple or zero mac codesigning identities
BUG=2206
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2012004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4527 4adac7df-926f-26a2-2b94-8c16560cd09d
uild/common.gypi
ibjingle_examples.gyp
|
| 91053e7c5a743f4a92f5079844b0747c927f3bbd |
10-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50654631.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2000006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/portallocatorfactory.cc
pp/webrtc/test/fakedtlsidentityservice.h
pp/webrtc/test/fakemediastreamsignaling.h
pp/webrtc/webrtc.scons
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
pp/webrtc/webrtcsessiondescriptionfactory.cc
pp/webrtc/webrtcsessiondescriptionfactory.h
ase/nssidentity.cc
ase/socket_unittest.cc
ase/sslidentity.cc
ase/sslidentity.h
ase/sslidentity_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/codec.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/webrtc/fakewebrtcvoiceengine.h
2p/base/basicpacketsocketfactory.cc
2p/base/constants.cc
2p/base/constants.h
2p/base/dtlstransport.h
2p/base/p2ptransportchannel_unittest.cc
2p/base/packetsocketfactory.h
2p/base/port.h
2p/base/portallocator.h
2p/base/session.cc
2p/base/session.h
2p/base/sessionmanager.cc
2p/base/transport.cc
2p/base/transport.h
2p/base/turnport.cc
2p/base/turnport_unittest.cc
2p/base/turnserver.cc
2p/client/basicportallocator.h
ession/media/mediasessionclient.cc
ession/media/mediasessionclient_unittest.cc
|
| 825e9b0a9b638aff124c8f79e7ac081ecb0df2d1 |
07-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk/objc/README: s/libjingle/webrtc/ in repository path.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1985004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4501 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/README
|
| c883fdc2737557b4b8db7686c76b9cf8f19a18fc |
06-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnection.java: enable setting trace & log levels from Java
Replaces the hard-coded scheme that was there before and lets apps decide what
to log and to where.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4498 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/Logging.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
ibjingle.gyp
|
| 9dba52562725dbaced0d671982201ede753d72e8 |
05-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
* Update libjingle to 50389769.
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004
RISK=P1
TESTED=try bots
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1967004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnectioninterface_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/nullvideoframe.h
edia/base/videoframe.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtctexturevideoframe.cc
edia/webrtc/webrtctexturevideoframe.h
edia/webrtc/webrtctexturevideoframe_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoframe.h
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/sessionmanager.cc
2p/base/sessionmanager.h
ession/media/call.cc
ession/media/call.h
ession/media/channelmanager.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient.h
ession/media/srtpfilter.cc
ession/media/srtpfilter.h
mpp/constants.cc
mpp/constants.h
mpp/mucroomdiscoverytask.cc
mpp/mucroomdiscoverytask.h
mpp/mucroomdiscoverytask_unittest.cc
mpp/mucroomlookuptask.h
|
| c3d93c692169121fc815cd32ac3527de64c4af89 |
05-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk/PRESUBMIT: Accept copyright years going back to 2004.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1956004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4485 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| a054569c15631aeab407fba92d782f3410cb3ef4 |
02-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix memory leak in datachannel and its test.
RISK=P3
TESTED=memcheck build
tools/valgrind-webrtc/webrtc_tests.sh --tool memcheck --test out/Debug/libjingle_peerconnection_unittest --gtest_filter=SctpDataChannelTest*
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1941005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4470 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel_unittest.cc
|
| 0dc0f172a3c3e2d6524ae4b67c0eafb1f661bbb2 |
01-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
sscanf isn't safe with strings that aren't null-terminated. In such case, create a local copy that is null-terminated first.
TESTED=GYP_DEFINES=build_for_tool=memcheck gclient runhooks
ninja -C out/Debug/ libjingle_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool memcheck --test out/Debug/libjingle_unittest --gtest_filter=Http*
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/1941004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4469 4adac7df-926f-26a2-2b94-8c16560cd09d
ase/httpbase.cc
|
| 86d7a198ec832b3f59bcb2baae18798b57bdee7e |
01-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
ObjC PeerConnection README: note workaround needed for crbug.com/248168
BUG=2106
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1940004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4467 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/README
|
| 1bc19541748f0b10e7b7d0eda732fee5a4389547 |
01-Aug-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo: builds using ninja on iOS for simulator and device!
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
the hand-crafted Xcode project (which has never worked in its checked-in
form), including a gyp action to sign the sample app for deployment to an iOS
device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
(in a surprising twist of fate, the API landed quite a bit later than the
sample app and this is the first time the CR-time changes in the API are
reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
formatting craxy.
BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1874005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/objc/README
pp/webrtc/objc/RTCICECandidate+Internal.h
pp/webrtc/objc/RTCICECandidate.mm
pp/webrtc/objc/RTCICEServer+Internal.h
pp/webrtc/objc/RTCICEServer.mm
pp/webrtc/objc/RTCIceCandidate+Internal.h
pp/webrtc/objc/RTCIceCandidate.mm
pp/webrtc/objc/RTCIceServer+Internal.h
pp/webrtc/objc/RTCIceServer.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/public/RTCICECandidate.h
pp/webrtc/objc/public/RTCICEServer.h
pp/webrtc/objc/public/RTCIceCandidate.h
pp/webrtc/objc/public/RTCIceServer.h
pp/webrtc/objc/public/RTCPeerConnectionFactory.h
pp/webrtc/objctests/README
uild/common.gypi
xamples/ios/AppRTCDemo.xcodeproj/project.pbxproj
xamples/ios/AppRTCDemo/APPRTCAppClient.h
xamples/ios/AppRTCDemo/APPRTCAppClient.m
xamples/ios/AppRTCDemo/APPRTCAppDelegate.h
xamples/ios/AppRTCDemo/APPRTCAppDelegate.m
xamples/ios/AppRTCDemo/AppRTCDemo-Info.plist
xamples/ios/AppRTCDemo/Info.plist
xamples/ios/AppRTCDemo/ResourceRules.plist
xamples/ios/README
xamples/ios/makeLibs.sh
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
|
| d64719d8954262fee94e7615422f3d027dc1ae6b |
01-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50191337.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1885005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannel_unittest.cc
pp/webrtc/datachannelinterface.h
ase/host.cc
ase/host.h
ase/host_unittest.cc
ase/httpcommon.cc
ase/nat_unittest.cc
ase/network.cc
ase/testclient_unittest.cc
ase/thread_unittest.cc
ibjingle.gyp
ibjingle_tests.gyp
edia/base/capturemanager.cc
edia/base/capturemanager_unittest.cc
edia/base/fakemediaengine.h
edia/base/mediachannel.h
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine_unittest.cc
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
2p/base/port_unittest.cc
2p/base/relayserver_unittest.cc
2p/base/session_unittest.cc
2p/base/stunserver_main.cc
2p/client/basicportallocator.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channelmanager.h
|
| 7fdbb1c832844694a62c9ff2a365f9f36f800d5f |
01-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
We don't need to link with libssl.so when we already depend on openssl.
This fixes the hidden-symbol linker warnings.
BUG=2149
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1927004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4459 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle.gyp
|
| caa7024b8616a0f22aeba0c54dd6ff0c4722600e |
31-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest.java: build on android bots as well as linux ones.
BUG=1796
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1921005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4455 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 85f07f59ee879dfcaa28ce63f0c831376167bffc |
30-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest.java: use java_home gyp var instead of hardcoding /usr.
BUG=1796
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1899005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4433 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_tests.gyp
|
| 3d496fb046bca52b8f5c9c194033dab1cdd550c4 |
30-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll chromium_revision 205140:214260 to pick up build fixes for ninja iOS device build.
TESTED=git try
BUG=2106
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1888005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4431 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 963856434072d1f67af31cadcb2baaad31f5a688 |
30-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds no parent to talk folder.
BUG=1933
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1896004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4430 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| d6134c7cfd809fd9e557899778eacc7d7c2728a6 |
29-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
PeerConnectionTest.java: make the test work for the bots' v4l2loopback.
- Make the test agnostic to the actual resolution used, since v4l2_file_player
is playing a non-640x480 file (go/httfw)
- Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what
v4l2_file_player is feeding.
Requires https://gist.github.com/fischman/2e9a9b2efd2ad363ef82 be applied to the
v4l2loopback driver code.
BUG=1796
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1891004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4422 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
|
| 147d44a4507851752182084bbf152358503755fa |
29-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo: replace the use of query-string parameters for pre-JB devices.
Replaces the use of a query-string parameter with a (once-per-session)
JS-to-Java function call, because query-string parameters on file:// URLs are
busted on ICS and earlier Android releases
(https://code.google.com/p/android/issues/detail?id=17535).
Also added channel.html to the list of inputs to cause edits to it to cause a
rebuild of the .apk.
BUG=1949
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1890004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4421 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/assets/channel.html
xamples/android/src/org/appspot/apprtc/GAEChannelClient.java
ibjingle_examples.gyp
|
| ea40bd0cc88855719f78903b2bed58550e19f8f5 |
29-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Presubmit script for preventing changes to protected files and add the full list of those files.
BUG=2090
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1855004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4419 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
|
| 1e09a711263dd105e6f7a03812250084c64e5fd8 |
26-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49952949
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
pp/webrtc/audiotrack.cc
pp/webrtc/audiotrack.h
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/firewallsocketserver.cc
ase/sslidentity_unittest.cc
ase/virtualsocketserver.cc
edia/base/audiorenderer.h
edia/base/capturerenderadapter.cc
edia/base/constants.cc
edia/base/constants.h
edia/base/fakemediaengine.h
edia/base/filemediaengine.cc
edia/base/filemediaengine.h
edia/base/mediachannel.h
edia/base/rtpdataengine.cc
edia/base/videoengine_unittest.h
edia/sctp/sctpdataengine.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcmediaengine.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvoiceengine.cc
edia/webrtc/webrtcvoiceengine.h
edia/webrtc/webrtcvoiceengine_unittest.cc
2p/base/dtlstransportchannel_unittest.cc
2p/base/port_unittest.cc
2p/base/portallocator.cc
2p/base/pseudotcp_unittest.cc
2p/base/relayserver.cc
2p/base/relayserver.h
2p/base/session.cc
2p/base/session_unittest.cc
2p/base/transport.h
2p/base/transportdescription.h
2p/client/connectivitychecker.cc
ession/media/call.cc
ession/media/channel.cc
ession/media/channel.h
ession/media/channel_unittest.cc
ession/media/mediamessages_unittest.cc
ession/media/mediasession.cc
ession/media/mediasessionclient.cc
ession/media/mediasessionclient_unittest.cc
ite_scons/site_tools/talk_linux.py
|
| c46967dc53a5f54e0126ba0fe5fdeafd9b584a38 |
25-Jul-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4391 "Roll chromium 205140:212975 to support ninja iOS ar..."
r4391 results in Mac Release Bot fail: http://chromegw/i/internal.client.webrtc/builders/Mac32%20Release/builds/334/steps/modules_integrationtests
> Roll chromium 205140:212975 to support ninja iOS armv7 build.
>
> In particular, picks up new clang, libvpx, libsrtp, yasm, and gyp.
>
> TESTED=git try on patchset #1
> BUG=2106
> R=henrike@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1849005
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1874004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4399 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 33584f942c6e1723918d9d2b76f429ab8396751e |
25-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Introduce a bit of sanity to talk/PRESUBMIT.py's license checking.
The comma this allows is a very common variant of the license header (3:1
preferred over the no-comma variant in talk/).
Also pacify pylint a bit, and correct a flagrantly incorrect header I happened
to come across.
BUG=2098,2133
R=henrike@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1866004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4396 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/peerconnectionfactory_unittest.cc
ite_scons/site_tools/talk_noops.py
ite_scons/talk.py
|
| 9fbc558dd408851e015cf3ecad3e3bca967ae8ed |
25-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk/OWNERS: add libjingle team members from internal webrtc/files/OWNERS
BUG=1933
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1867004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4395 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| 880c8426274720e04db45320ee2416a33edcc7bf |
24-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
AppRTCDemo: don't render frames that are already outdated.
BUG=2121
R=henrike@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1850004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4392 4adac7df-926f-26a2-2b94-8c16560cd09d
xamples/android/src/org/appspot/apprtc/VideoStreamsView.java
|
| 87f8a7eb67e2948208a422c9e2d0cc45e7f3489e |
24-Jul-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll chromium 205140:212975 to support ninja iOS armv7 build.
In particular, picks up new clang, libvpx, libsrtp, yasm, and gyp.
TESTED=git try on patchset #1
BUG=2106
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4391 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
|
| 8d27a1c7238fc66fad6b1928b7f92947eb0bfc38 |
23-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
BUG=1932
TESTED=git try
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1851004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4385 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_all.gyp
|
| 5c280ecd5762494809cd5ade5e018bb867a01fc9 |
23-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4382 "Makes webrtc and libjingle build from the same gyp-..."
Failures: breaks build bots. Will have to disable Android NDK build for libjingle. The TSAN issues are in webrtc which should be unaffected. Flakey? Here are the failing tests:
http://chromegw/i/internal.client.webrtc/builders/Android%20NDK/builds/303 and http://chromegw/i/internal.client.webrtc/builders/Linux%20Tsan/builds/284
> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
>
> BUG=1932
> TESTED=git try
> R=andrew@webrtc.org, fischman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1836004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1834005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4383 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_all.gyp
|
| 5fcddf2334832c8fed3f45621175ba48ef1c0580 |
23-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
BUG=1932
TESTED=git try
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1836004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4382 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_all.gyp
|
| 390fcb7a202163a22ae13979a0246f66180b2011 |
23-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
BUG=2091
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1833004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4381 4adac7df-926f-26a2-2b94-8c16560cd09d
RESUBMIT.py
pp/webrtc/OWNERS
|
| 28654cbc2256230c978f41cbaf550bc2e9c2f2db |
22-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49713299.
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1848004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/common.cc
ase/common.h
ase/helpers.cc
ase/md5.cc
ase/sslstreamadapter_unittest.cc
uild/common.gypi
xamples/peerconnection/client/main_wnd.cc
xamples/peerconnection/client/peer_connection_client.cc
xamples/peerconnection/server/data_socket.cc
ibjingle.gyp
edia/base/capturerenderadapter.cc
edia/base/fakenetworkinterface.h
edia/base/fakevideocapturer.h
edia/base/filemediaengine_unittest.cc
edia/base/hybridvideoengine.h
edia/base/mediachannel.h
edia/base/mediaengine.h
edia/base/testutils.cc
edia/base/videoadapter.cc
edia/base/videoadapter.h
edia/base/videoengine_unittest.h
edia/base/videoframe.cc
edia/devices/fakedevicemanager.h
edia/devices/filevideocapturer.cc
edia/sctp/sctpdataengine.cc
edia/sctp/sctpdataengine.h
edia/webrtc/fakewebrtcdeviceinfo.h
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
edia/webrtc/webrtcpassthroughrender.h
edia/webrtc/webrtcvideoengine.cc
edia/webrtc/webrtcvideoengine.h
edia/webrtc/webrtcvideoengine_unittest.cc
edia/webrtc/webrtcvideoframe.cc
edia/webrtc/webrtcvoiceengine.cc
2p/base/dtlstransportchannel_unittest.cc
2p/base/fakesession.h
2p/base/port.cc
2p/base/port_unittest.cc
2p/base/portallocatorsessionproxy.cc
2p/base/portallocatorsessionproxy_unittest.cc
2p/base/pseudotcp.cc
2p/base/pseudotcp_unittest.cc
2p/base/relayport.cc
2p/base/relayserver.cc
2p/base/relayserver_unittest.cc
2p/base/session_unittest.cc
2p/base/stun.cc
2p/base/stun_unittest.cc
2p/base/stunserver_unittest.cc
2p/base/turnport.cc
2p/base/turnserver.cc
ession/media/channel.cc
ession/media/channel_unittest.cc
ession/media/channelmanager.cc
ession/media/channelmanager.h
ession/media/channelmanager_unittest.cc
ession/media/mediasession.cc
ession/media/mediasession_unittest.cc
mpp/jid.cc
mpp/xmppclient.cc
|
| 0df5b8dfa6a26c34f9b9e1091407dcf509db6267 |
18-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4372 "Makes webrtc and libjingle build from the same gyp-..."
> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
>
> TESTED=git try
> BUG=1932
> R=andrew@webrtc.org, fischman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1804004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1835004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4373 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_all.gyp
|
| 4e4bf4db8bbbfda5d98f89d8475b9762afa2a8c8 |
18-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
TESTED=git try
BUG=1932
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1804004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4372 4adac7df-926f-26a2-2b94-8c16560cd09d
ibjingle_all.gyp
|
| 8c7347124c2760cbf1340759253e7f046490d10e |
17-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
talk: DataChannel.java repeated contents. This removes the duplicate.
TBR=ajm
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1825004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4365 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/src/org/webrtc/DataChannel.java
|
| 9de257d00f1f805af28f15fd814a8a84460028e5 |
17-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.
TBR=wu@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1824004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/webrtcsession_unittest.cc
uild/common.gypi
ibjingle.gyp
ibjingle.scons
ibjingle_tests.gyp
edia/base/mediachannel.h
edia/base/rtpdataengine_unittest.cc
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/webrtcvoiceengine.cc
2p/base/tcpport.cc
2p/client/portallocator_unittest.cc
ession/media/channel.h
ession/media/channel_unittest.cc
|
| 7b2f955e565a64d182b25ae3763b09b60fc682b8 |
16-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.
BUG=1935
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1805005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
OPYING
ICENSE_THIRD_PARTY
|
| 723d683ecbe6a934885a60712c66ca2c01700a51 |
12-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1797004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
pp/webrtc/OWNERS
pp/webrtc/datachannelinterface.h
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/DataChannel.java
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/portallocatorfactory.cc
pp/webrtc/webrtcsession_unittest.cc
ase/httpbase_unittest.cc
ase/multipart_unittest.cc
ase/nat_unittest.cc
ase/network.cc
ase/network_unittest.cc
ase/physicalsocketserver_unittest.cc
ase/proxydetect.cc
ase/signalthread_unittest.cc
ase/socket_unittest.cc
uild/common.gypi
xamples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
ibjingle.gyp
ibjingle_examples.gyp
ibjingle_tests.gyp
edia/webrtc/fakewebrtcvideoengine.h
edia/webrtc/fakewebrtcvoiceengine.h
ession/media/channelmanager.cc
ession/media/channelmanager.h
|
| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
pp/webrtc/audiotrack.cc
pp/webrtc/audiotrack.h
pp/webrtc/audiotrackrenderer.cc
pp/webrtc/audiotrackrenderer.h
pp/webrtc/datachannel.cc
pp/webrtc/datachannel.h
pp/webrtc/datachannelinterface.h
pp/webrtc/dtmfsender.cc
pp/webrtc/dtmfsender.h
pp/webrtc/dtmfsender_unittest.cc
pp/webrtc/dtmfsenderinterface.h
pp/webrtc/fakeportallocatorfactory.h
pp/webrtc/java/README
pp/webrtc/java/jni/peerconnection_jni.cc
pp/webrtc/java/src/org/webrtc/AudioSource.java
pp/webrtc/java/src/org/webrtc/AudioTrack.java
pp/webrtc/java/src/org/webrtc/IceCandidate.java
pp/webrtc/java/src/org/webrtc/MediaConstraints.java
pp/webrtc/java/src/org/webrtc/MediaSource.java
pp/webrtc/java/src/org/webrtc/MediaStream.java
pp/webrtc/java/src/org/webrtc/MediaStreamTrack.java
pp/webrtc/java/src/org/webrtc/PeerConnection.java
pp/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
pp/webrtc/java/src/org/webrtc/SdpObserver.java
pp/webrtc/java/src/org/webrtc/SessionDescription.java
pp/webrtc/java/src/org/webrtc/StatsObserver.java
pp/webrtc/java/src/org/webrtc/StatsReport.java
pp/webrtc/java/src/org/webrtc/VideoCapturer.java
pp/webrtc/java/src/org/webrtc/VideoRenderer.java
pp/webrtc/java/src/org/webrtc/VideoSource.java
pp/webrtc/java/src/org/webrtc/VideoTrack.java
pp/webrtc/javatests/libjingle_peerconnection_java_unittest.sh
pp/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
pp/webrtc/jsep.h
pp/webrtc/jsepicecandidate.cc
pp/webrtc/jsepicecandidate.h
pp/webrtc/jsepsessiondescription.cc
pp/webrtc/jsepsessiondescription.h
pp/webrtc/jsepsessiondescription_unittest.cc
pp/webrtc/localaudiosource.cc
pp/webrtc/localaudiosource.h
pp/webrtc/localaudiosource_unittest.cc
pp/webrtc/localvideosource.cc
pp/webrtc/localvideosource.h
pp/webrtc/localvideosource_unittest.cc
pp/webrtc/mediaconstraintsinterface.cc
pp/webrtc/mediaconstraintsinterface.h
pp/webrtc/mediastream.cc
pp/webrtc/mediastream.h
pp/webrtc/mediastream_unittest.cc
pp/webrtc/mediastreamhandler.cc
pp/webrtc/mediastreamhandler.h
pp/webrtc/mediastreamhandler_unittest.cc
pp/webrtc/mediastreaminterface.h
pp/webrtc/mediastreamprovider.h
pp/webrtc/mediastreamproxy.h
pp/webrtc/mediastreamsignaling.cc
pp/webrtc/mediastreamsignaling.h
pp/webrtc/mediastreamsignaling_unittest.cc
pp/webrtc/mediastreamtrack.h
pp/webrtc/mediastreamtrackproxy.h
pp/webrtc/notifier.h
pp/webrtc/objc/README
pp/webrtc/objc/RTCAudioTrack+Internal.h
pp/webrtc/objc/RTCAudioTrack.mm
pp/webrtc/objc/RTCEnumConverter.h
pp/webrtc/objc/RTCEnumConverter.mm
pp/webrtc/objc/RTCI420Frame.mm
pp/webrtc/objc/RTCIceCandidate+Internal.h
pp/webrtc/objc/RTCIceCandidate.mm
pp/webrtc/objc/RTCIceServer+Internal.h
pp/webrtc/objc/RTCIceServer.mm
pp/webrtc/objc/RTCMediaConstraints+Internal.h
pp/webrtc/objc/RTCMediaConstraints.mm
pp/webrtc/objc/RTCMediaConstraintsNative.cc
pp/webrtc/objc/RTCMediaConstraintsNative.h
pp/webrtc/objc/RTCMediaSource+Internal.h
pp/webrtc/objc/RTCMediaSource.mm
pp/webrtc/objc/RTCMediaStream+Internal.h
pp/webrtc/objc/RTCMediaStream.mm
pp/webrtc/objc/RTCMediaStreamTrack+Internal.h
pp/webrtc/objc/RTCMediaStreamTrack.mm
pp/webrtc/objc/RTCPair.m
pp/webrtc/objc/RTCPeerConnection+Internal.h
pp/webrtc/objc/RTCPeerConnection.mm
pp/webrtc/objc/RTCPeerConnectionFactory.mm
pp/webrtc/objc/RTCPeerConnectionObserver.h
pp/webrtc/objc/RTCPeerConnectionObserver.mm
pp/webrtc/objc/RTCSessionDescription+Internal.h
pp/webrtc/objc/RTCSessionDescription.mm
pp/webrtc/objc/RTCVideoCapturer+Internal.h
pp/webrtc/objc/RTCVideoCapturer.mm
pp/webrtc/objc/RTCVideoRenderer+Internal.h
pp/webrtc/objc/RTCVideoRenderer.mm
pp/webrtc/objc/RTCVideoSource+Internal.h
pp/webrtc/objc/RTCVideoSource.mm
pp/webrtc/objc/RTCVideoTrack+Internal.h
pp/webrtc/objc/RTCVideoTrack.mm
pp/webrtc/objc/public/RTCAudioSource.h
pp/webrtc/objc/public/RTCAudioTrack.h
pp/webrtc/objc/public/RTCI420Frame.h
pp/webrtc/objc/public/RTCIceCandidate.h
pp/webrtc/objc/public/RTCIceServer.h
pp/webrtc/objc/public/RTCMediaConstraints.h
pp/webrtc/objc/public/RTCMediaSource.h
pp/webrtc/objc/public/RTCMediaStream.h
pp/webrtc/objc/public/RTCMediaStreamTrack.h
pp/webrtc/objc/public/RTCPair.h
pp/webrtc/objc/public/RTCPeerConnection.h
pp/webrtc/objc/public/RTCPeerConnectionDelegate.h
pp/webrtc/objc/public/RTCPeerConnectionFactory.h
pp/webrtc/objc/public/RTCSessionDescription.h
pp/webrtc/objc/public/RTCSessionDescriptonDelegate.h
pp/webrtc/objc/public/RTCTypes.h
pp/webrtc/objc/public/RTCVideoCapturer.h
pp/webrtc/objc/public/RTCVideoRenderer.h
pp/webrtc/objc/public/RTCVideoRendererDelegate.h
pp/webrtc/objc/public/RTCVideoSource.h
pp/webrtc/objc/public/RTCVideoTrack.h
pp/webrtc/objctests/Info.plist
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.h
pp/webrtc/objctests/RTCPeerConnectionSyncObserver.m
pp/webrtc/objctests/RTCPeerConnectionTest.mm
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.h
pp/webrtc/objctests/RTCSessionDescriptionSyncObserver.m
pp/webrtc/objctests/mac/main.mm
pp/webrtc/peerconnection.cc
pp/webrtc/peerconnection.h
pp/webrtc/peerconnection_unittest.cc
pp/webrtc/peerconnectionfactory.cc
pp/webrtc/peerconnectionfactory.h
pp/webrtc/peerconnectionfactory_unittest.cc
pp/webrtc/peerconnectioninterface.h
pp/webrtc/peerconnectioninterface_unittest.cc
pp/webrtc/peerconnectionproxy.h
pp/webrtc/portallocatorfactory.cc
pp/webrtc/portallocatorfactory.h
pp/webrtc/proxy.h
pp/webrtc/proxy_unittest.cc
pp/webrtc/statscollector.cc
pp/webrtc/statscollector.h
pp/webrtc/statscollector_unittest.cc
pp/webrtc/statstypes.h
pp/webrtc/streamcollection.h
pp/webrtc/test/fakeaudiocapturemodule.cc
pp/webrtc/test/fakeaudiocapturemodule.h
pp/webrtc/test/fakeaudiocapturemodule_unittest.cc
pp/webrtc/test/fakeconstraints.h
pp/webrtc/test/fakeperiodicvideocapturer.h
pp/webrtc/test/fakevideotrackrenderer.h
pp/webrtc/test/mockpeerconnectionobservers.h
pp/webrtc/test/testsdpstrings.h
pp/webrtc/videosourceinterface.h
pp/webrtc/videosourceproxy.h
pp/webrtc/videotrack.cc
pp/webrtc/videotrack.h
pp/webrtc/videotrack_unittest.cc
pp/webrtc/videotrackrenderers.cc
pp/webrtc/videotrackrenderers.h
pp/webrtc/webrtc.scons
pp/webrtc/webrtcsdp.cc
pp/webrtc/webrtcsdp.h
pp/webrtc/webrtcsdp_unittest.cc
pp/webrtc/webrtcsession.cc
pp/webrtc/webrtcsession.h
pp/webrtc/webrtcsession_unittest.cc
ase/asyncfile.cc
ase/asyncfile.h
ase/asynchttprequest.cc
ase/asynchttprequest.h
ase/asynchttprequest_unittest.cc
ase/asyncpacketsocket.h
ase/asyncsocket.cc
ase/asyncsocket.h
ase/asynctcpsocket.cc
ase/asynctcpsocket.h
ase/asynctcpsocket_unittest.cc
ase/asyncudpsocket.cc
ase/asyncudpsocket.h
ase/asyncudpsocket_unittest.cc
ase/atomicops.h
ase/atomicops_unittest.cc
ase/autodetectproxy.cc
ase/autodetectproxy.h
ase/autodetectproxy_unittest.cc
ase/bandwidthsmoother.cc
ase/bandwidthsmoother.h
ase/bandwidthsmoother_unittest.cc
ase/base64.cc
ase/base64.h
ase/base64_unittest.cc
ase/basicdefs.h
ase/basictypes.h
ase/basictypes_unittest.cc
ase/bind.h
ase/bind.h.pump
ase/bind_unittest.cc
ase/buffer.h
ase/buffer_unittest.cc
ase/bytebuffer.cc
ase/bytebuffer.h
ase/bytebuffer_unittest.cc
ase/byteorder.h
ase/byteorder_unittest.cc
ase/checks.cc
ase/checks.h
ase/common.cc
ase/common.h
ase/constructormagic.h
ase/cpumonitor.cc
ase/cpumonitor.h
ase/cpumonitor_unittest.cc
ase/crc32.cc
ase/crc32.h
ase/crc32_unittest.cc
ase/criticalsection.h
ase/cryptstring.h
ase/dbus.cc
ase/dbus.h
ase/dbus_unittest.cc
ase/diskcache.cc
ase/diskcache.h
ase/diskcache_win32.cc
ase/diskcache_win32.h
ase/event.cc
ase/event.h
ase/event_unittest.cc
ase/fakecpumonitor.h
ase/fakenetwork.h
ase/fakesslidentity.h
ase/faketaskrunner.h
ase/filelock.cc
ase/filelock.h
ase/filelock_unittest.cc
ase/fileutils.cc
ase/fileutils.h
ase/fileutils_mock.h
ase/fileutils_unittest.cc
ase/firewallsocketserver.cc
ase/firewallsocketserver.h
ase/flags.cc
ase/flags.h
ase/gunit.h
ase/gunit_prod.h
ase/helpers.cc
ase/helpers.h
ase/helpers_unittest.cc
ase/host.cc
ase/host.h
ase/host_unittest.cc
ase/httpbase.cc
ase/httpbase.h
ase/httpbase_unittest.cc
ase/httpclient.cc
ase/httpclient.h
ase/httpcommon-inl.h
ase/httpcommon.cc
ase/httpcommon.h
ase/httpcommon_unittest.cc
ase/httprequest.cc
ase/httprequest.h
ase/httpserver.cc
ase/httpserver.h
ase/httpserver_unittest.cc
ase/ifaddrs-android.cc
ase/ifaddrs-android.h
ase/ipaddress.cc
ase/ipaddress.h
ase/ipaddress_unittest.cc
ase/json.cc
ase/json.h
ase/json_unittest.cc
ase/latebindingsymboltable.cc
ase/latebindingsymboltable.cc.def
ase/latebindingsymboltable.h
ase/latebindingsymboltable.h.def
ase/latebindingsymboltable_unittest.cc
ase/libdbusglibsymboltable.cc
ase/libdbusglibsymboltable.h
ase/linked_ptr.h
ase/linux.cc
ase/linux.h
ase/linux_unittest.cc
ase/linuxfdwalk.c
ase/linuxfdwalk.h
ase/linuxfdwalk_unittest.cc
ase/linuxwindowpicker.cc
ase/linuxwindowpicker.h
ase/linuxwindowpicker_unittest.cc
ase/logging.cc
ase/logging.h
ase/logging_unittest.cc
ase/macasyncsocket.cc
ase/macasyncsocket.h
ase/maccocoasocketserver.h
ase/maccocoasocketserver.mm
ase/maccocoasocketserver_unittest.mm
ase/maccocoathreadhelper.h
ase/maccocoathreadhelper.mm
ase/macconversion.cc
ase/macconversion.h
ase/macsocketserver.cc
ase/macsocketserver.h
ase/macsocketserver_unittest.cc
ase/macutils.cc
ase/macutils.h
ase/macutils_unittest.cc
ase/macwindowpicker.cc
ase/macwindowpicker.h
ase/macwindowpicker_unittest.cc
ase/mathutils.h
ase/md5.cc
ase/md5.h
ase/md5digest.h
ase/md5digest_unittest.cc
ase/messagedigest.cc
ase/messagedigest.h
ase/messagedigest_unittest.cc
ase/messagehandler.cc
ase/messagehandler.h
ase/messagequeue.cc
ase/messagequeue.h
ase/messagequeue_unittest.cc
ase/multipart.cc
ase/multipart.h
ase/multipart_unittest.cc
ase/nat_unittest.cc
ase/natserver.cc
ase/natserver.h
ase/natserver_main.cc
ase/natsocketfactory.cc
ase/natsocketfactory.h
ase/nattypes.cc
ase/nattypes.h
ase/nethelpers.cc
ase/nethelpers.h
ase/network.cc
ase/network.h
ase/network_unittest.cc
ase/nssidentity.cc
ase/nssidentity.h
ase/nssstreamadapter.cc
ase/nssstreamadapter.h
ase/nullsocketserver.h
ase/nullsocketserver_unittest.cc
ase/openssladapter.cc
ase/openssladapter.h
ase/openssldigest.cc
ase/openssldigest.h
ase/opensslidentity.cc
ase/opensslidentity.h
ase/opensslstreamadapter.cc
ase/opensslstreamadapter.h
ase/optionsfile.cc
ase/optionsfile.h
ase/optionsfile_unittest.cc
ase/pathutils.cc
ase/pathutils.h
ase/pathutils_unittest.cc
ase/physicalsocketserver.cc
ase/physicalsocketserver.h
ase/physicalsocketserver_unittest.cc
ase/posix.cc
ase/posix.h
ase/profiler.cc
ase/profiler.h
ase/profiler_unittest.cc
ase/proxy_unittest.cc
ase/proxydetect.cc
ase/proxydetect.h
ase/proxydetect_unittest.cc
ase/proxyinfo.cc
ase/proxyinfo.h
ase/proxyserver.cc
ase/proxyserver.h
ase/ratelimiter.cc
ase/ratelimiter.h
ase/ratelimiter_unittest.cc
ase/ratetracker.cc
ase/ratetracker.h
ase/ratetracker_unittest.cc
ase/refcount.h
ase/referencecountedsingletonfactory.h
ase/referencecountedsingletonfactory_unittest.cc
ase/rollingaccumulator.h
ase/rollingaccumulator_unittest.cc
ase/schanneladapter.cc
ase/schanneladapter.h
ase/scoped_autorelease_pool.h
ase/scoped_autorelease_pool.mm
ase/scoped_ptr.h
ase/scoped_ref_ptr.h
ase/sec_buffer.h
ase/sha1.cc
ase/sha1.h
ase/sha1digest.h
ase/sha1digest_unittest.cc
ase/sharedexclusivelock.cc
ase/sharedexclusivelock.h
ase/sharedexclusivelock_unittest.cc
ase/signalthread.cc
ase/signalthread.h
ase/signalthread_unittest.cc
ase/sigslot.h
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mpp/xmpptask.cc
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