| 0e7e259ebd69993bb5670a991f43aa1b06c9bf9e |
|
13-Nov-2015 |
mflodman <mflodman@webrtc.org> |
Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
/external/webrtc/webrtc/video/video_send_stream.h
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=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 -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 373284da06631c33a01fee6d755355cabeadf6c6 |
|
03-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Make SendStatisticsProxy outlive ViEChannel.
Prevents data race on stats_proxy_ on VideoSendStream destruction.
BUG=
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1409133006 .
Cr-Commit-Position: refs/heads/master@{#10493}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 0c478b3d75be3c026e68f03a11cb558c3655c926 |
|
21-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Rename ChannelGroup to CongestionController and move to webrtc/call/.
BUG=webrtc:5079
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1419803002 .
Cr-Commit-Position: refs/heads/master@{#10358}
/external/webrtc/webrtc/video/video_send_stream.h
|
| e37870297fc45f1253dff4b1c59c85a50d2a8a97 |
|
21-Oct-2015 |
mflodman <mflodman@webrtc.org> |
ChannelGroup cleanup.
Move CallStats to Call, EncoderStateFeedback to VideoSendStream and
remove last ViEChannel dependency from ChannelGroup.
BUG=webrtc:5079
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1418613002 .
Cr-Commit-Position: refs/heads/master@{#10355}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 0dbf0090a961c5e5fb7362937108337564b4a91f |
|
19-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Remove the video channel id completely.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1412143002
Cr-Commit-Position: refs/heads/master@{#10324}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 949c2f04b4156095090e02f3f13613aadacce88d |
|
16-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Move ownership of send ViEChannels and ViEEncoder to VideoSendStream.
This is the first CL to get ready for adapting audio bitrate based on
BWE. I've kept this CL as small as possible and had to add a few getters
to ChannelManager. The next CL will do the same for receive ViEChannels.
The getters are a bit uggly, but is an in-between-state. Let's discuss
future ownership of the different modules and what do do with
ChannelGroup.
BUG=5079
Review URL: https://codereview.webrtc.org/1394243006
Cr-Commit-Position: refs/heads/master@{#10298}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 5c389d3e09646c0e2ed76d5ccb37a3419a09eb6a |
|
25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.
BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1227923005 .
Cr-Commit-Position: refs/heads/master@{#10073}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 45553aefacb797818da83ccef1c3679a8aa0fc7f |
|
08-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoEngine interface usage from new API.
Instantiates ProcessThread/ChannelGroup inside Call instead of using
VideoEngine or ViEBase. This removes the need for ViEChannelManager,
ViEInputManager and other ViESharedData completely.
Some interface headers are still referenced due to external interfaces
being defined there. Upon interface removal these will be moved to
implementation headers.
BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50849005
Cr-Commit-Position: refs/heads/master@{#9160}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 5cb9ce4c746867a02e7d37358f63e1a7c11ef262 |
|
05-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViECodec usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also inlining the necessary parts of SetSendCodec
done previously in ViECodecImpl.
BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46129004
Cr-Commit-Position: refs/heads/master@{#9136}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.h
|
| ddbddbdee626825f8c8e9a98fd1f8fce8f234ea9 |
|
28-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViENetwork usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder and ViEChannel
removing a layer of indirection.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50809004
Cr-Commit-Position: refs/heads/master@{#9096}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 038df3c5d7c77ae5a6047ef99da1ddf79597a985 |
|
28-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViEExternalCodec usage in VideoSendStream.
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49179004
Cr-Commit-Position: refs/heads/master@{#9095}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 59d91dc951143995069798edee05e757502f335c |
|
27-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViERTP_RTCP usage in VideoSendStream.
Replaces interface usage with direct calls on ChannelGroup, ViEEncoder
and ViEChannel removing a layer of indirection.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43319004
Cr-Commit-Position: refs/heads/master@{#9088}
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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}
/external/webrtc/webrtc/video/video_send_stream.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
/external/webrtc/webrtc/video/video_send_stream.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
/external/webrtc/webrtc/video/video_send_stream.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
/external/webrtc/webrtc/video/video_send_stream.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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 32452b20b8f5ea4470ec619a31eefc736e51c8a3 |
|
22-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make ReconfigureVideoEncoder use current bitrate.
Prevents bitrate drops when changing resolution etc.
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/24069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7493 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.h
|
| dde16f19e3ed36ca462f6404c40d5a9811f0ec37 |
|
06-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix some code styles.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22009004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 168f23faa5b8a49d4dd709c6649e77d5fecf36bf |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 4ef438e2defd6c46404f6b367287364cde66b7fb |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 2bb1bdab8d11f5445693c028335fb3ace631f636 |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Preserve RTP states for restarted VideoSendStreams.
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| be9d2a45499d87f3b04e644fc173b0d997a9eeea |
|
30-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reserve RTP/RTCP modules in SetSSRC.
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.
Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.
To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.
BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| eb16b811fb5c21954f9e4a6f4c64e297e9fd65b9 |
|
16-Jun-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implements start bitrate for new video API.
Added a new rampup test.
BUG=2879
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 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
/external/webrtc/webrtc/video/video_send_stream.h
|
| 1566ee289364fdac5aa9dcc62db3070033208ad1 |
|
23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
This reverts commit r6230 to re-land r6229.
ViECapturer::SwapFrame now resets timestamps.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 2cdd433edfec8d02c5f49fc634a8a07fc7e792ca |
|
23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Remove VideoSendStreamInput::PutFrame."
This reverts r6229.
Test WebRtcVideoChannel2BaseTest.MuteStream fails after r6229.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6230 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| f3085e43ab7e0f6cb8c89cb02ed9e5694aba2e96 |
|
23-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove VideoSendStreamInput::PutFrame.
PutFrame just copies the frame before swapping it, if it's required that
can easily be done outside this API before swapping the frame.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6229 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| de1429e9ad9a3a207ca191e1d748aa7271066860 |
|
28-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add thread annotations to Call API.
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| a5c8d2c9b39a2d20fead2147e60ed0cd6d62019c |
|
24-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename Start/Stop in Video{Send,Receive}Streams.
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.
R=mflodman@webrtc.org
BUG=3227
Review URL: https://webrtc-codereview.appspot.com/12329005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| f577ae9eac9822380ea6f0fb953cf383d0ec5374 |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove internal codecs from VideoSendStream.
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings
struct. The EncoderSettings struct uses an external encoder for all
codecs. This means that external users, such as libjingle, will provide
the encoders themselves, removing the previous distinction of internal
and external codecs.
For now VideoSendStream translates to VideoCodec internally. In the
interrim (before the corresponding change is implemented in
VideoReceiveStream) tests convert EncoderSettings to VideoCodecs.
Removes Call::GetVideoCodecs().
Disables RampUpTest.WithPacingAndRtx as its further exposed with changes
to bitrates used in tests.
BUG=2854,2992
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 09315705b9caf3bff455e3515b9bd99492a7c3e3 |
|
07-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up statistics in video receive stream of new API
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| bdc5ed2e7d081de6597f0f993d54489eb3abd496 |
|
31-Jan-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add configuration for cpu overuse detection to video send stream.
BUG=2422
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5468 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| ccd42840bcee8db145be91b3308912a24f710a6f |
|
07-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up statistics in video send stream of new video engine api
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5559006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| f3973e81d5aa7e4f1d6b5abdfe3a6dc53a32840c |
|
13-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure channels in the same call are in the same channel group.
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 724947b8efa44d15d699b471020005450590f5b6 |
|
11-Dec-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add SwapFrame() to VideoSendStreamInput.
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 4070935f4fb5b9fb2df246d7073fe0ba7e350791 |
|
26-Nov-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement and test EncodedImageCallback in new ViE API.
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 53c85735256dc7d540deb0a5e2bbb2f2821c4bd4 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename video streams' start/stop methods.
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 6488761f2e6ce7b977bbc14bc7b91933527d633a |
|
14-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement VideoSendStream::SetCodec().
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|
| 16e03b7bd8b88ba569987e20a7f29061f91a3d0d |
|
28-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/video_send_stream.h
|