After https://codereview.webrtc.org/2386573002 changed where resolution
changes are handled, a few VideoSendStreamTests became flaky. They were
waiting for an InitEncode call they triggered, but sometimes were
getting one triggered by the resolution change when the first frame was
generated.
The fix was to make the tests wait for two InitEncode calls first -
one when the stream is created, and the second when the first frame was
encoded.
BUG=webrtc:6467, webrtc:6371
R=perkj@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2387293002 .
Cr-Commit-Position: refs/heads/master@{#14490}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
The implementation is borrowed from Chromium.
Also change use of raw pointers in VideoSendStreamImpl to use WeakPtr<>
BUG= webrtc:6289
Review-Url: https://codereview.webrtc.org/2367853002
Cr-Commit-Position: refs/heads/master@{#14468}
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.
Original cl description:
Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2378103005 .
Cr-Commit-Position: refs/heads/master@{#14452}
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
Reason for revert:
Caused issues with webrtc_perf_tests on build bots.
Original issue's description:
> Fix race / crash in OnNetworkRouteChanged().
>
> To achieve this some refactoring was done to make it possible to synchronize
> access to the DelayBasedBwe in TransportFeedbackAdapter:
> - The callback was removed from DelayBasedBwe, it now instead returns its
> result.
> - TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
> unnecessary dependencies.
>
> Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
>
> BUG=webrtc:6427, webrtc:6422
>
> Committed: https://crrev.com/fd0d42669204e6dd92a60736bca7ae0196663024
> Cr-Commit-Position: refs/heads/master@{#14430}
TBR=terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6427, webrtc:6422
Review-Url: https://codereview.webrtc.org/2377303002
Cr-Commit-Position: refs/heads/master@{#14433}
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.
BUG=webrtc:6371
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/2368223002 .
Cr-Commit-Position: refs/heads/master@{#14431}
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
Review-Url: https://codereview.webrtc.org/2366333003
Cr-Commit-Position: refs/heads/master@{#14430}
This is done to ensure GN targets are placed in the same directory as of the source files.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
Reason for revert:
Fix backward compatibility support
Original issue's description:
> Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
>
> Reason for revert:
> Breaks compilation of internal downstream project.
>
> Original issue's description:
> > Unify rtcp packet setters
> > Renamed setters in rtcp classes
> > from WithField to SetField
> > from WithItem to AddItem or SetItems
> > from From to SetSenderSsrc
> > from To to SetMediaSsrc
> > Some redundant or unsued setters removed.
> > Pass-by-const& replaced with pass-by-value when appropriate.
> >
> > BUG=webrtc:5260
> >
> > Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> > Cr-Commit-Position: refs/heads/master@{#14393}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5260
>
> Committed: https://crrev.com/efc6e41866662e0922858fbce1d9ee3bdd0637ed
> Cr-Commit-Position: refs/heads/master@{#14400}
TBR=sprang@webrtc.org,stefan@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2370313002
Cr-Commit-Position: refs/heads/master@{#14402}
Reason for revert:
Breaks compilation of internal downstream project.
Original issue's description:
> Unify rtcp packet setters
> Renamed setters in rtcp classes
> from WithField to SetField
> from WithItem to AddItem or SetItems
> from From to SetSenderSsrc
> from To to SetMediaSsrc
> Some redundant or unsued setters removed.
> Pass-by-const& replaced with pass-by-value when appropriate.
>
> BUG=webrtc:5260
>
> Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> Cr-Commit-Position: refs/heads/master@{#14393}
TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2372713005
Cr-Commit-Position: refs/heads/master@{#14400}
Original description:
Add proper lifetime of encoder-specific settings.
Permits passing VideoEncoderConfig between threads and not worry about
the lifetime of an underlying void pointer. Also adds type safety to
unpacking of codec-specific settings.
These settings are not yet propagating to VideoEncoder interfaces, but
the aim is to get rid of webrtc::VideoCodec for VideoEncoder.
BUG=webrtc:3424
R=perkj@webrtc.org, pbos@webrtc.orgTBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2347843002
Cr-Commit-Position: refs/heads/master@{#14396}
Renamed setters in rtcp classes
from WithField to SetField
from WithItem to AddItem or SetItems
from From to SetSenderSsrc
from To to SetMediaSsrc
Some redundant or unsued setters removed.
Pass-by-const& replaced with pass-by-value when appropriate.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2348623003
Cr-Commit-Position: refs/heads/master@{#14393}
This cl change so that all encoder configuration changes are reported to VideoSendStream through the ViEEncoder.
Also, the PayLoadRouter is changed to never stop sending on a an ssrc due to the encoder video frame size changes. Instead, the number of sending streams is only decided by the number of sending ssrc.
This cl is a preparation for moving encoder reconfiguration due to input video frame size changes from WebRtcVideoSendStream to ViEEncoder.
BUG=webrtc:5687, webrtc:6371
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/2338133003 .
Cr-Commit-Position: refs/heads/master@{#14371}
"WebRTC.Video.EndToEndDelayInMs"
Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).
BUG=webrtc:6409
Review-Url: https://codereview.webrtc.org/1905563002
Cr-Commit-Position: refs/heads/master@{#14367}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
Deleted from the VideoFrameBuffer base class.
BUG=webrtc:5921
Review-Url: https://codereview.webrtc.org/2278883002
Cr-Commit-Position: refs/heads/master@{#14317}
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values
This cl
Revert "Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ )"
This reverts commit 9fdbda6aa3f66ea872344c22e79b23361047cbab.
and fix the problem in the original cl in video_quality_test.cc
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2348533002
Cr-Commit-Position: refs/heads/master@{#14265}
Reason for revert:
Fails on Mac and Linux webrtc_perf_tests
Original issue's description:
> Replace VideoCapturerInput with VideoSinkInterface.
> Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
>
> This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
>
> BUG=webrtc:5687
> // Android CQ seems broken.
> NOTRY=true
>
> Committed: https://crrev.com/95a226f55ae7e32b83a6ba96232fb105a014dc6c
> Cr-Commit-Position: refs/heads/master@{#14238}
TBR=nisse@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2344923002
Cr-Commit-Position: refs/heads/master@{#14239}
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
BUG=webrtc:5687
// Android CQ seems broken.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2257413002
Cr-Commit-Position: refs/heads/master@{#14238}
Integrate AvgCounter to be used for BWE stats in call.
Fixes for stats regression in:
WebRTC.Call.EstimatedSendBitrateInKbps
WebRTC.Call.PacerBitrateInKbps
Example:
BWE for a 15 seconds long call (with intervals of 1 sec):
|300|400|500|600|600|600|600| 0 | 0 | 0 | 0 | 0 |800|800|800| // 0 - network state down
Reported via OnNetworkChanged:
|300|400|500|600| x | x | x | 0 | x | x | x | x |800| x | x | // x - empty interval, 0 -> pauses stats
Stats:
|300|400|500|600|600|600|600| - | - | - | - | - |800|800|800| // x -> last value used (intervals during pause ignored)
AvgCounter uses the average of samples within an interval (interval length is 2 sec).
BUG=webrtc:6244
Review-Url: https://codereview.webrtc.org/2307913002
Cr-Commit-Position: refs/heads/master@{#14147}
Reason for revert:
Breaks webrtc_perf_tests on Windows, Mac and Linux (that test don't run on trybots):
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/8841/steps/webrtc_perf_tests/logs/stdio
Example:
[ RUN ] FullStackTest.ForemanCifWithoutPacketLossVp9
# Fatal error in ../../webrtc/video/video_quality_test.cc, line 1056
# last system error: 34
# Check failed: !params_.audio.enabled
Original issue's description:
> Separating video settings in VideoQualityTest.
>
> This is a simple refactoring of VideoQualityTest. It will help in adding audio related settings to VideoQualityTest.
>
> BUG=
>
> Committed: https://crrev.com/f07fb0013164bdb031dcc88dc83365a27643b2d9
> Cr-Commit-Position: refs/heads/master@{#14139}
TBR=stefan@webrtc.org,minyue@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2325723002
Cr-Commit-Position: refs/heads/master@{#14142}
This is a simple refactoring of VideoQualityTest. It will help in adding audio related settings to VideoQualityTest.
BUG=
Review-Url: https://codereview.webrtc.org/2312613003
Cr-Commit-Position: refs/heads/master@{#14139}
Reason for revert:
Downstream build is fixed.
Original issue's description:
> Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
>
> Reason for revert:
> Breaks downstream build.
>
> Original issue's description:
> > Ignore Camera and Flip bits in CVO when parsing video rotation
> >
> > Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
> > set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
> > The Camera and Flip bit is still unimplemented and will just be ignored
> > though.
> >
> > BUG=webrtc:6120
> > R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
> >
> > Committed: f9e1b922ef
>
> TBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6120
>
> Committed: https://crrev.com/97667c7746282704acccd896e26175decee349c0
> Cr-Commit-Position: refs/heads/master@{#14035}
TBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6120
Review-Url: https://codereview.webrtc.org/2320913003
Cr-Commit-Position: refs/heads/master@{#14124}
"WebRTC.Call.BitrateReceivedInKbps"
"WebRTC.Call.VideoBitrateReceivedInKbps"
"WebRTC.Call.AudioBitrateReceivedInKbps"
"WebRTC.Call.RtcpBitrateReceivedInBps"
Reports the average of periodically computed bitrates over a call (2 sec interval). Adds possibility to later modify the stats and use/report the periodic bitrates.
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2303763002
Cr-Commit-Position: refs/heads/master@{#14119}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
This test failed on the memcheck bot:
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/6704/steps/video_engine_tests/logs/stdio
The test assumed that the absolute send time header extension can never
be zero. It's a timestamp truncated to 24 bits, and zero is not a
special value - so it can very rarely end up being precisely zero.
The fix makes the test wait for at least one packet having a non-zero send time.
I've considered changing the test to use a fake clock instead to ensure
that not only the value is non-zero, but that it indeed reflects the
system timestamp - but that involves changing a very large number of
files. Besides, other tests in this file don't verify values for header
extensions where zeroes are allowed.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2307693002
Cr-Commit-Position: refs/heads/master@{#14056}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
Reason for revert:
Breaks downstream build.
Original issue's description:
> Ignore Camera and Flip bits in CVO when parsing video rotation
>
> Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
> set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
> The Camera and Flip bit is still unimplemented and will just be ignored
> though.
>
> BUG=webrtc:6120
> R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
>
> Committed: f9e1b922efTBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6120
Review-Url: https://codereview.webrtc.org/2300323002
Cr-Commit-Position: refs/heads/master@{#14035}
Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
The Camera and Flip bit is still unimplemented and will just be ignored
though.
BUG=webrtc:6120
R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2280703002 .
Cr-Commit-Position: refs/heads/master@{#14027}
cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/
- Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
TBR=mflodman@webrtc.org
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
