| 04cb763955aadf29324ec124b55aa399a7645d51 |
|
14-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Add tests for verifying transport feedback for audio and video.
BUG=webrtc:5263
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1589523002 .
Cr-Commit-Position: refs/heads/master@{#11255}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| e74eef19bd3f101208dc72b98038e42fc523a351 |
|
08-Jan-2016 |
stefan <stefan@webrtc.org> |
Add CreateSend/ReceiveTransport() methods to CallTest.
This allows the test to create its own transports if it, for instance, needs to do demuxing.
BUG=webrtc:5416
Review URL: https://codereview.webrtc.org/1573453002
Cr-Commit-Position: refs/heads/master@{#11187}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 9fea80f50daab46f20d4a6fc67b0144fbbbf56cd |
|
07-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Add audio streams to CallTest and a first A/V call test.
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.
Audio streams are using a fake audio device with file input.
The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.
R=pbos@webrtc.org
TBR=kjellander@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1542653002 .
Cr-Commit-Position: refs/heads/master@{#11171}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| e2976c87f7ba627fa1e1246f0ccfb34b4b9f3a73 |
|
04-Jan-2016 |
Peter Boström <pbos@webrtc.org> |
Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| ff483617a4fdf282bb82d7f4ce15af3dbe305a4a |
|
21-Dec-2015 |
stefan <stefan@webrtc.org> |
Step 1 to prepare call_test.* for combined audio/video tests.
Also move (and clean up includes) rampup_tests.* to webrtc/call in preparation for combined audio/video ramp-up tests.
No functional changes.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1537273003
Cr-Commit-Position: refs/heads/master@{#11101}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 5811a39f14fd77ebc0793ee93d03ee15a669bd8f |
|
10-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Replace EventWrapper in video/, test/ and call/.
Makes use of rtc::Event which is simpler and can be used without
allocating additional objects on the heap.
Does not modify test/channel_transport/.
BUG=
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1487893004 .
Cr-Commit-Position: refs/heads/master@{#10968}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| d1590b2571c4cb33416e14c92e4f2dfed42ec3d4 |
|
09-Dec-2015 |
mflodman <mflodman@webrtc.org> |
Lint clean video/ and add lint presubmit check.
BUG=webrtc:5316
Review URL: https://codereview.webrtc.org/1507643004
Cr-Commit-Position: refs/heads/master@{#10953}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 5f6deaf52531e69ea94750b3403fbdb228208b8a |
|
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Remove unused RTP-header parser.
D'oh.
BUG=
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1506743003 .
Cr-Commit-Position: refs/heads/master@{#10915}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 03671cb38a53b7d4f22545f5e58079a234f7cf27 |
|
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Use existing parser in ReceivesAndRetransmitsNack.
Removes logspam of "Failed to find extension id:".
BUG=
TBR=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1502993003 .
Cr-Commit-Position: refs/heads/master@{#10913}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| b4a1ae5299fd57be66c7cbb7a982179bb1ecfb90 |
|
03-Dec-2015 |
sprang <sprang@webrtc.org> |
Add separate send-side UMA stats for screenshare and video.
This CL duplicates all the histograms in SendStatisticsProxy. Might be
overkill, but we don't know which stats will be interesting and it makes
the change easier.
BUG=
Review URL: https://codereview.webrtc.org/1433393002
Cr-Commit-Position: refs/heads/master@{#10885}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| db81ffd6f50d487441555947ec63ccc766e75043 |
|
23-Nov-2015 |
jbauch <jbauch@webrtc.org> |
Request keyframe if too many packets are missing and NACK is disabled.
This allows enabling "EndToEndTest.ReceivesPliAndRecoversWithoutNack".
BUG=webrtc:2250
Review URL: https://codereview.webrtc.org/1211873004
Cr-Commit-Position: refs/heads/master@{#10747}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 18adf0a79d4a0ea124c7f27fd553382d0b0cb7ac |
|
17-Nov-2015 |
stefan <stefan@webrtc.org> |
Add UMA for send bwe and pacer bitrate.
Review URL: https://codereview.webrtc.org/1434403004
Cr-Commit-Position: refs/heads/master@{#10675}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 6f14be8df8b67feb480f55b3a41e2b8cd06a836d |
|
16-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Add limit for minimum number of required samples before recording input and sent framerate stats.
BUG=
Review URL: https://codereview.webrtc.org/1446443002
Cr-Commit-Position: refs/heads/master@{#10644}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 91d926038f7cebf889ef843f2f087d72bc8c60c2 |
|
11-Nov-2015 |
stefan <stefan@webrtc.org> |
Add receive bitrate UMA stats.
Review URL: https://codereview.webrtc.org/1440603002
Cr-Commit-Position: refs/heads/master@{#10605}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 608213e737054e2b15ea51f37a11bd5b48e914aa |
|
01-Nov-2015 |
stefan <stefan@webrtc.org> |
Add locks and thread annotations for ReceiverReferenceTimeReportEnabled.
Review URL: https://codereview.webrtc.org/1413543007
Cr-Commit-Position: refs/heads/master@{#10473}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
|
27-Oct-2015 |
stefan <stefan@webrtc.org> |
Call OnSentPacket for all packets sent in the test framework.
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 65e7d4cf20c32f6ecd0dcb9c0db0ac94ed8aafbe |
|
26-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Remove CanCreateAndDestroyManyVideoStreams.
This test was used to verify that VideoEngine handles were handed back
correctly. This is no longer applicable.
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1425673002 .
Cr-Commit-Position: refs/heads/master@{#10412}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 861c55e58311383b7f4f61af463ddea53eb3f30f |
|
16-Oct-2015 |
sprang <sprang@webrtc.org> |
Transport sequence number should be set also for retransmissions.
This is a reland of https://codereview.webrtc.org/1385563005 which was
reverted since the test was flaky. The reason was a race condition (in
the test) and that NACK wasn't properly set up.
BUG=
Review URL: https://codereview.webrtc.org/1406193002
Cr-Commit-Position: refs/heads/master@{#10303}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 10950692d67af5cfdcb19d93b40f25193d1db8c6 |
|
06-Oct-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Transport sequence number should be set also for retransmissions."
After this CL, video_engine_test started failing flakily in different bots for different CLs.
TBR=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1393553003 .
Cr-Commit-Position: refs/heads/master@{#10188}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| af4ced986bc62c263fbdb6eab68aef5c0d4e7c78 |
|
06-Oct-2015 |
sprang <sprang@webrtc.org> |
Transport sequence number should be set also for retransmissions.
When fetching a packet from the rtp packet history, cuased by a
retransmission, the transport seq extension header is enabled but the
sequence number is set to 0. A new transport seq should be assigned in
this case.
BUG=
Review URL: https://codereview.webrtc.org/1385563005
Cr-Commit-Position: refs/heads/master@{#10183}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 6b8d3551681f40b880507cecc88f478a12383cc7 |
|
24-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/
The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc
BUG=webrtc:4173
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1362303002 .
Cr-Commit-Position: refs/heads/master@{#10052}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| c9bbeb03542cffc14b7d306e5f88b6c0e593864d |
|
23-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29
Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}
TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1362923002 .
Cr-Commit-Position: refs/heads/master@{#10029}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| ef165eefc79cf28bb67779afe303cc2365885547 |
|
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Wire up send-side bandwidth estimation.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1338203003
Cr-Commit-Position: refs/heads/master@{#10012}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 8d62971611bac3e19d4489427a4b8346ad1e4ca8 |
|
04-Aug-2015 |
Erik Språng <sprang@webrtc.org> |
Fix race condition in EndToEndTest.AssignsTransportSequenceNumbers
Don't verify increasing sequence numbers after test complesion as this
can be racy with regards to test shutting down send transports.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1269743004 .
Cr-Commit-Position: refs/heads/master@{#9672}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 867fb5224e1ba6a1c2cd523c005499a93ed61a08 |
|
03-Aug-2015 |
sprang <sprang@webrtc.org> |
Add support for transport wide sequence numbers
Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 6718e97e730dfeb0c4290128b5682e123dd75866 |
|
24-Jul-2015 |
asapersson <asapersson@webrtc.org> |
Add encode and decode time to histograms stats:
- "WebRTC.Video.EncodeTimeInMs"
- "WebRTC.Video.DecodeTimeInMs"
BUG=chromium:488243
Review URL: https://codereview.webrtc.org/1250203002
Cr-Commit-Position: refs/heads/master@{#9630}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| d89920b74a173b7bf80c6760908a382c095a66cc |
|
22-Jul-2015 |
asapersson <asapersson@webrtc.org> |
Add resolution and fps stats to histograms:
- "WebRTC.Video.InputWidthInPixels"
- "WebRTC.Video.InputHeightInPixels"
- "WebRTC.Video.SentWidthInPixels"
- "WebRTC.Video.SentHeightInPixels"
- "WebRTC.Video.ReceivedWidthInPixels"
- "WebRTC.Video.ReceivedHeightInPixels"
- "WebRTC.Video.RenderFramesPerSecond"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1228393008
Cr-Commit-Position: refs/heads/master@{#9611}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| a7d70546ad40f535b76b1ef57afc28bcba8ce09f |
|
07-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove VCM_*_PAYLOAD_TYPE constants.
These payload types aren't directly connected to any payload type, and
the payload type still has to be negotiated externally. As such these
constants are just a source of confusion.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1215603003
Cr-Commit-Position: refs/heads/master@{#9546}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 24b4eda6f4fdfd33d2c3e82df1390bad55953f5d |
|
16-Jun-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add sent framerates to histogram stats:
"WebRTC.Video.InputFramesPerSecond",
"WebRTC.Video.SentFramesPerSecond".
BUG=488243
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1169543005.
Cr-Commit-Position: refs/heads/master@{#9446}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| def39883f00c25525dfd34c3cee92b428e54b9e5 |
|
27-May-2015 |
Peter Boström <pbos@webrtc.org> |
Configure default render delay as 10 ms.
BUG=chromium:488395
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56419005
Cr-Commit-Position: refs/heads/master@{#9296}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 242e22b055940be70b1df3031e2363b0d02397b2 |
|
11-May-2015 |
Erik Språng <sprang@webrtc.org> |
Refactor RTCP sender
The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but
it has quite a few ramifications. Notable changes:
* Removed the rtcpPacketTypeFlags bit vector and don't assume
RTCPPacketType values have a single unique bit set. This will allow
making this an enum class once rtcp_receiver has been overhauled.
* Flags are now stored in a map that is a member of the class. This
meant we could remove some bool flags (eg send_remb_) which was
previously masked into rtcpPacketTypeFlags and then masked out again
when testing if a remb packet should be sent.
* Make all build methods, eg. BuildREMB(), have the same signature.
An RtcpContext struct was introduced for this purpose. This allowed
the use of a map from RTCPPacketType to method pointer. Instead of
18 consecutive if-statements, there is now a single loop.
The context class also allowed some simplifications in the build
methods themselves.
* A few minor simplifications and cleanups.
The next step is to gradually replace the builder methods with the
builders from the new RtcpPacket classes.
BUG=2450
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48329004
Cr-Commit-Position: refs/heads/master@{#9166}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 01b488831bf7cb3276d8bdfbe0204dfbdbbba725 |
|
05-May-2015 |
Stefan Holmer <stefan@webrtc.org> |
Use padding to achieve bitrate probing if the initial key frame has too few packets.
BUG=4350
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44879004
Cr-Commit-Position: refs/heads/master@{#9134}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| f2f828374c3ee1e1834c72bb27eaae88ef67bb40 |
|
01-May-2015 |
Peter Boström <pbos@webrtc.org> |
Use rtc::CriticalSection in webrtc/video/.
Removes heap allocation from CriticalSection creation.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50839004
Cr-Commit-Position: refs/heads/master@{#9126}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 3c391cbabb5416ca7275d3f6e6cc4fb49c4cf523 |
|
27-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api.
Add tests for verifying that video histograms are updated.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44309004
Cr-Commit-Position: refs/heads/master@{#9085}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| db313b667a1ba7986aa87fab0f65b74e2de7b17b |
|
02-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Disable EndToEndTest.ReceivedFecPacketsNotNacked on all platforms.
The test seems to flake on all platforms.
See webrtc:4328 for more info.
BUG=4328
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43029004
Cr-Commit-Position: refs/heads/master@{#8919}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| f7b9cf54a6b6085b67696d8d9c3c53b0b67758f5 |
|
30-Mar-2015 |
Minyue Li <minyue@webrtc.org> |
Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan
BUG=4328
R=kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47859004
Cr-Commit-Position: refs/heads/master@{#8892}
/external/webrtc/webrtc/video/end_to_end_tests.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}
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 2b4ce3a501b8d679f84c1ad10317dea5c78fa595 |
|
23-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Convert webrtc/video/ abort/assert to CHECK/DCHECK.
Also replaces NULL with nullptr. This gives nicer error messages and
keeps style consistent.
BUG=1756
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42879004
Cr-Commit-Position: refs/heads/master@{#8831}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8831 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 5c928ebd1d5d61deca0b5458e31005ac5dc1c31f |
|
25-Feb-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Let first packet through to avoid getting key frame requests (and no nacks) for EndToEndTest.ReceivedFecPacketsNotNacked.
BUG=4328
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38259004
Cr-Commit-Position: refs/heads/master@{#8502}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8502 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 8278c072b69c17c5c1d1d4880894152e9a8355ac |
|
23-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Enable NACK under SendsAndReceivesH264.
Decoding with errors has a bug that triggers an assert during packet
loss. Switching to NACK since that is what we expected to be running.
BUG=4337
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43399004
Cr-Commit-Position: refs/heads/master@{#8458}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8458 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 131bea89d6f3742e649be84c91f8fd6c43b62d28 |
|
18-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Offline screenshare quality test, plus loopback.
BUG=4171
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34109004
Cr-Commit-Position: refs/heads/master@{#8408}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| d5ce2e63dfef7d3744f16ef23174aef23dd72e8f |
|
13-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove EventWrapper::Reset().
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41939004
Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 37c0559c1edd108b345abcce1939f9b8d78d02a3 |
|
28-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets).
Don't copy codec specific header for empty packets in the jitter buffer.
BUG=3135
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37659004
Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| ba253473da5e8c9622504c353ea5de08aac2fa41 |
|
25-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Reenable GetStats test.
Also increasing start bitrate to have the test go significantly faster
on average. Hopefully an assert hit in the jitter buffer while running
this test was fixed in r7735.
R=stefan@webrtc.org
BUG=4014
Review URL: https://webrtc-codereview.appspot.com/26239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7744 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 67c22478a4a4aa9307cf1538815294a4a99e95f5 |
|
14-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Disable EndToEnd.GetStats test.
Looks like this test exposes a bug in jitter buffer after enabling
multiple streams. Will disable to be able to debug it in peace and not
have to revert.
TBR=stefan@webrtc.org
BUG=4014
Review URL: https://webrtc-codereview.appspot.com/31009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7704 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| a9c2d454bd0b544a8f5c3e3c2e8f695a4418b473 |
|
13-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix and enable CanReceiveFec test.
Test relied on the first protected media packet that was dropped to
actually be rendered, while rendering it could have been skipped on slow
systems due to newer frames being decoded before rendering happens.
R=stefan@webrtc.org
BUG=3269
Review URL: https://webrtc-codereview.appspot.com/25159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7696 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 5f1e2e42a8f8198016f5a965ea3113adfcce19cf |
|
06-Nov-2014 |
marpan@webrtc.org <marpan@webrtc.org> |
Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 09cc686c8bc4b5e766d38e6860ac52aa886e2436 |
|
04-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Delete VideoReceiveStream channels in destructor.
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/31909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7611 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 5b8831782074d490969171de5f8c67251f36d9cc |
|
01-Nov-2014 |
marpan@webrtc.org <marpan@webrtc.org> |
Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932
R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| ad3b5a5c16ff768def84138147d592ecb669a8cd |
|
24-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Move min transmit bitrate to VideoEncoderConfig.
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| b1dac33cac5a64cbec6b0fd72624fa9d3060376c |
|
17-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
BUG=3932
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/27779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 4ddbbed16ec1dad823d9a7fc1b0e2eecc4c36e31 |
|
10-Oct-2014 |
marpan@webrtc.org <marpan@webrtc.org> |
Disable SendsAndReceivesVP9 test for now.
Fails on linux memcheck and DrMemory.
Will re-enable on next libvpx roll.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7424 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 573c78e31c7ccdc5cf44ebc54b9fc089f5e8f0cf |
|
10-Oct-2014 |
marpan@webrtc.org <marpan@webrtc.org> |
Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.
R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 02686115ccb25583668da001aabce34efde75536 |
|
19-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Re-enable missing android tests disabled due to issue 3770.
BUG=3770
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7238 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 01581da71145d4b9504d12cfad0c988d1fc68654 |
|
04-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Fix audio/video sync when FEC is enabled.
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.
BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 6f729e8a74a4990ca2560607cbc9907cdfaf0401 |
|
02-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Disable video_engine_tests and webrtc_perf_tests on Android.
BUG=3770
TESTED=Running the tests locally on an Android device.
R=phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| b623c5c251d9a6372aff1d65dbeec8274cac08ea |
|
26-Aug-2014 |
aluebs@webrtc.org <aluebs@webrtc.org> |
Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky
BUG=webrtc:3745
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6981 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 79c3359e67bdbf3394a19f9251927c9482174043 |
|
06-Aug-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add end-to-end H.264 packetization test.
Also correctly wires up H.264 packetization in the new Call api.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| f9460688a61ccac0067feef07192e05a44e5d7e3 |
|
24-Jul-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure padding is sent on the first sending RTP module.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 62bafae6618fe3aefbd18657062abc98a40c3375 |
|
08-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some refactoring inside rtp_rtcp/.
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.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
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 20c1f56992ae14cc75b8ae81e8eb3f9a99db0c1e |
|
04-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Configure RTX send status on new modules.
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).
As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|
| 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/end_to_end_tests.cc
|
| 994d0b7229a18b255d81979c2bedaf8ecfae9bd7 |
|
27-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor Call-based tests.
Greatly reduces duplication of constants and setup code for tests based
on the new webrtc::Call APIs. It also makes it significantly easier to
convert sender-only to end-to-end tests as they share more code.
BUG=3035
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/video/end_to_end_tests.cc
|