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ef53a7f
Reset IO thread checker when iOS audio unit stops
by Brian Dai
· 2 hours ago
master
bf95da8
Update WebRTC code version (2021-01-23T04:04:31).
by webrtc-version-updater
· 6 hours ago
8c007ff
Restrict usage of resolution bitrate limits to singlecast
by Sergey Silkin
· 14 hours ago
lkgr
461b1d9
Restart CPU overuse detection when encoder settings has changed.
by Jakob Ivarsson
· 19 hours ago
2803a2d
Make audio device mocks publicly visible
by Steve Anton
· 19 hours ago
8df643b
Introduce FinalRefCountedObject template class
by Danil Chapovalov
· 20 hours ago
cbacec5
Monitor the "concealed samples" stat for the audio during negotiation.
by Harald Alvestrand
· 23 hours ago
11215fe
Require scalability mode to initialize av1 encoder.
by Danil Chapovalov
· 26 hours ago
2ed56fe
Update WebRTC code version (2021-01-22T04:03:26).
by webrtc-version-updater
· 30 hours ago
d2dd732
Introduce network emulated endpoint optional name for better logging
by Artem Titov
· 2 days ago
e4fd1ba
Delete mutable rtc::CopyOnWriteBuffer::data
by Danil Chapovalov
· 2 days ago
6031b74
Implement a Neon optimized function to find the argmax element in an array.
by Ivo Creusen
· 2 days ago
03eed7c
Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial.
by Erik Språng
· 2 days ago
a7e34d3
Add resolution_bitrate_limits to EncoderInfo field trial.
by Åsa Persson
· 2 days ago
026ad9a
Update WebRTC code version (2021-01-21T04:03:14).
by webrtc-version-updater
· 2 days ago
49b20f9
Fix race with SctpTransport destruction and usrsctp timer thread.
by Taylor Brandstetter
· 2 days ago
c2ae4c8
Allow separate dump sets for the data dumper in APM
by Per Åhgren
· 3 days ago
0be1846
Fix enabling DependencyDescriptor for VP9 with spatial layers
by Danil Chapovalov
· 3 days ago
1657baf
Add `.cache` to .gitignore.
by Rasmus Brandt
· 3 days ago
4f281f1
Cleanup FakeRtcEventLog from thread awareness
by Danil Chapovalov
· 3 days ago
812c73c
Another ilbc cross correlation fix
by Ivo Creusen
· 3 days ago
5eb43b4
Prefix HAVE_SCTP macro with WEBRTC_.
by Mirko Bonadei
· 3 days ago
6dcbcea
Update WebRTC code version (2021-01-20T04:04:26).
by webrtc-version-updater
· 3 days ago
33c0ab4
Call MediaChannel::OnPacketReceived on the network thread.
by Tomas Gunnarsson
· 4 days ago
1cbf21e
ChannelStatistics RTT test case around remote SSRC change.
by Tim Na
· 4 days ago
a24d35e
AlignmentAdjuster: take reduced layers into account for default downscaling.
by Åsa Persson
· 4 days ago
5c3ff6b
Switch to enable the HMM transparent mode classifier
by Gustaf Ullberg
· 4 days ago
801c999
Signal extmap-allow-mixed by default on session level
by Emil Lundmark
· 4 days ago
1e75df2
Remove lock from UlpfecReceiverImpl and replace with a sequence checker.
by Tomas Gunnarsson
· 4 days ago
5eb527c
Replace sigslot usages with callback list library.
by Lahiru Ginnaliya Gamathige
· 4 days ago
a722d2a
Add DeliverPacketAsync method to PacketReceiver.
by Tomas Gunnarsson
· 4 days ago
6cdb67f
Document expected thread safety of the TaskQueue interface
by Danil Chapovalov
· 4 days ago
676d61f
Update WebRTC code version (2021-01-19T04:01:32).
by webrtc-version-updater
· 4 days ago
29bd863
Add field trial for allowing cropped resolution when limiting max layers.
by Åsa Persson
· 5 days ago
5cf0ef0
Stricter compile-time thread annotations in JsepTransportController
by Niels Möller
· 5 days ago
25b8235
Remove unused function VideoDecoder::PrefersLateDecoding.
by philipel
· 5 days ago
42eef86
Remove unused code in APM
by Alessio Bazzica
· 5 days ago
111a371
Delete unused.h include from api as unused
by Danil Chapovalov
· 5 days ago
3e9cb2c
Move deprecated code to their own build targets.
by Niels Möller
· 5 days ago
844c759
fix variable naming in ReportSdpFormatReceived
by Philipp Hancke
· 5 days ago
ee95c1c
Roll chromium_revision da24822732..18311e2720 (844357:844473)
by chromium-webrtc-autoroll
· 5 days ago
77ceff9
payload type mapper: use media constants
by Philipp Hancke
· 5 days ago
4bab23f
Update pc/ to use C++ lambdas instead of rtc::Bind
by Niels Möller
· 5 days ago
d51000f
Delete RTC_WARN_UNUSED_RESULT as no longer used
by Danil Chapovalov
· 5 days ago
cc6ae44
Reland "Improve structuring of test for audio glitches."
by Harald Alvestrand
· 5 days ago
c20e333
Update SCTP test to use C++ lambdas instead of rtc::Bind
by Niels Möller
· 5 days ago
588526c
Remove deprecated //rtc_base:async_resolver.
by Mirko Bonadei
· 5 days ago
e091fd2
Remove lock from RtpStreamReceiverController.
by Tomas Gunnarsson
· 5 days ago
8467cf2
Reduce redundant flags for audio stream playout state.
by Tomas Gunnarsson
· 5 days ago
12971a2
Update WebRTC code version (2021-01-18T04:03:42).
by webrtc-version-updater
· 5 days ago
09729d2
Update WebRTC code version (2021-01-17T04:01:53).
by webrtc-version-updater
· 6 days ago
9c8dd87
Fixing WebRTC/Chromium FYI build.
by Mirko Bonadei
· 7 days ago
be93b78
Move iOS bundle data for tests inside rtc_include_test (take 2).
by Mirko Bonadei
· 7 days ago
2397b6e
SimulcastEncoderAdapter: Add field trial for EncoderInfo settings.
by Åsa Persson
· 7 days ago
fd9500e
In criket::BaseChannel replace AsyncInvoker with task queue functions
by Danil Chapovalov
· 7 days ago
8ed6185
Move iOS bundle data for tests inside rtc_include_test.
by Mirko Bonadei
· 7 days ago
d705bbe
Roll chromium_revision 3c2d1e3ba1..da24822732 (844134:844357)
by chromium-webrtc-autoroll
· 7 days ago
f4d1ecf
Update WebRTC code version (2021-01-16T04:03:13).
by webrtc-version-updater
· 7 days ago
61ede7a
Roll chromium_revision 72fe6d7aab..3c2d1e3ba1 (844008:844134)
by chromium-webrtc-autoroll
· 8 days ago
d723da1
Reland "Default enable delay adaptation during DTX."
by Jakob Ivarsson
· 8 days ago
098da17
Reland "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code"
by Danil Chapovalov
· 8 days ago
ba91dbc
In SVC controllers add support for frames dropped by encoder
by Danil Chapovalov
· 8 days ago
e5f4c6b
Reland "Refactor rtc_base build targets."
by Mirko Bonadei
· 8 days ago
79d9c37
Revert "Default enable delay adaptation during DTX."
by Jakob Ivarsson
· 8 days ago
59bdcbe
Default enable delay adaptation during DTX.
by Jakob Ivarsson
· 8 days ago
5ab6a8c
Refactors SimulcastEncoder Adapter.
by Erik Språng
· 8 days ago
1528e2b
Set AV1E_SET_ERROR_RESILIENT_MODE on T1 and T2 enhanced layers
by Sergio Garcia Murillo
· 8 days ago
809a261
Roll chromium_revision 42ab9dc8c8..72fe6d7aab (843550:844008)
by chromium-webrtc-autoroll
· 8 days ago
a86cef7
Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding
by Danil Chapovalov
· 8 days ago
87e9f6e
Update p2p/ to use C++ lambdas instead of rtc::Bind
by Niels Möller
· 8 days ago
3ae09f5
Revert "Improve structuring of test for audio glitches."
by Alex Loiko
· 8 days ago
98db5d1
Revert "Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll"
by Alex Loiko
· 8 days ago
b45d3aa
Update android jni code to use C++ lambdas instead of rtc::Bind
by Niels Möller
· 8 days ago
d76dcbd
Simplify FakeRtcEventLog, delete rtc::Bind usage
by Niels Möller
· 8 days ago
f9ee0e0
Add cross trafic emulation api
by Andrey Logvin
· 8 days ago
a7ccacc
Update WebRTC code version (2021-01-15T04:04:08).
by webrtc-version-updater
· 8 days ago
6c8738c
mb: Fully remove references to 'masters' in favor of 'builder_groups'.
by Mirko Bonadei
· 9 days ago
7acc2d9
Revert "Refactor rtc_base build targets."
by Mirko Bonadei
· 9 days ago
884118d
Delete unused functions in ModuleRtpRtcpImpl
by Danil Chapovalov
· 9 days ago
23f60eb
Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll
by Austin Orion
· 9 days ago
c17bca7
SetOfferedRtpHeaderExtensions: fix error code.
by Markus Handell
· 9 days ago
1669103
Roll chromium_revision 189823ba75..42ab9dc8c8 (843026:843550)
by chromium-webrtc-autoroll
· 9 days ago
f77aa81
Update AudioDeviceBuffer to use C++ lambdas instead of rtc::Bind
by Niels Möller
· 9 days ago
42c0d70
Include packetization in video codec string
by Emil Lundmark
· 9 days ago
4319b16
Revert "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code"
by Danil Chapovalov
· 9 days ago
8c2250e
Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
by Danil Chapovalov
· 9 days ago
1921708
SetNegotiatedHeaderExtensions_w: Set list synchronously.
by Markus Handell
· 9 days ago
c12f625
Adds VideoDecoder::GetDecoderInfo()
by Erik Språng
· 9 days ago
a0bb2ef
Delete unused VideoType enum values
by Niels Möller
· 9 days ago
e61a40e
Fix typo in audio processing header.
by Hua, Chunbo
· 9 days ago
fdbaeda
Improve structuring of test for audio glitches.
by Harald Alvestrand
· 9 days ago
dbcaff0
Fix AudioProcessing::Config::ToString() implementation.
by Hua, Chunbo
· 9 days ago
d73426d6
Add new empty build targets rtp_rtcp_legacy and video_legacy.
by Niels Möller
· 9 days ago
b24e720
Fix inconsistencies in network BUILD.gn file
by Andrey Logvin
· 9 days ago
db79204
Change PeerConnectionE2EQualityTest to use lambdas instead of rtc::Bind
by Niels Möller
· 9 days ago
76714a6
AGC2 minor code clean up
by Alessio Bazzica
· 9 days ago
ece6712
Add av1 to lower range IDs.
by Jerome Jiang
· 10 days ago
507eacf
Reland "ChannelStatistics used for RTP stats in VoipStatistics."
by Tim Na
· 10 days ago
2297272
Roll chromium_revision c0feffab8f..189823ba75 (842900:843026)
by chromium-webrtc-autoroll
· 10 days ago
8606b9c
Replace all uses of the word 'master' with 'builder_group' in //tools/mb
by Mirko Bonadei
· 10 days ago
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