- b55c299 Avoid TBR in DEPS autoroller (use CC instead). by Mirko Bonadei · 2 years, 6 months ago
- e204466 Add unit test of AudioRtpReceiver by Tony Herre · 2 years, 6 months ago
- 2db59a6 dcsctp: Use rtc::CopyOnWriteBuffer by Victor Boivie · 2 years, 7 months ago
- 8695282 Remove unnecessary copy of suspended_ssrcs. by Tommi · 2 years, 6 months ago
- 9345bee DEPS: use python3 by Takuto Ikuta · 2 years, 6 months ago
- ef5b21e Deprecate and remove usage for WARNING log level by Harald Alvestrand · 2 years, 6 months ago
- 1505035 Delete unused classes Win32SocketServer, Win32Socket, Win32Thread. by Niels Möller · 2 years, 6 months ago
- b2860b0 Update peerconnection example to not use Win32Socket by Niels Möller · 2 years, 9 months ago
- deb1b1b Always call IsOk() to ensure audio codec configuration is valid when negotiating. by Ivo Creusen · 2 years, 6 months ago
- 789a0f3 Delete deprecated RtpExtension::FindHeaderExtensionByUri variant by Danil Chapovalov · 2 years, 6 months ago
- a40e6de Allow extremely low resolution for simulcast path by Ilya Nikolaevskiy · 2 years, 6 months ago
- e1bbef1 Add options to only NACK if there is a valid RTT and if loss rate is below a configured value. by Jakob Ivarsson · 2 years, 6 months ago
- 144e5bf Use NONE if scalability mode is not specified by Sergey Silkin · 2 years, 6 months ago
- 61a8d9c Call: Deduplicate SentPacket notifications by Jianhui Dai · 2 years, 6 months ago
- 5f34130 Declare LERROR deprecated and remove all usage in webrtc by Harald Alvestrand · 2 years, 6 months ago
- 4e8e36c Try to fix the version updated bot. by Mirko Bonadei · 2 years, 6 months ago
- a018e67 Optimize block_delay_buffer. by cschuldt · 2 years, 6 months ago
- 624fb67 Revert "Fix out-of-bounds memory access due to large number of audio channels." by Ivo Creusen · 2 years, 6 months ago
- c3a1ea1 dcsctp: assert that CreateTimeout returns a usable pointer by Philipp Hancke · 2 years, 6 months ago
- a18cad9 Add compile-time warning for RTC_LOG(INFO) by Harald Alvestrand · 2 years, 6 months ago
- 6c3c3db Roll chromium_revision 866506397b..1ce7d592c3 (944390:944520) by chromium-webrtc-autoroll · 2 years, 6 months ago
- b09d872 Reland "Add dav1d decoder to WebRTC." by philipel · 2 years, 6 months ago
- 707e5a0 Make test framework create portallocator with an explicit PacketSocketFactory. by Niels Möller · 2 years, 6 months ago
- aad4ba1 Fix windows bot command line issue. by Jeremy Leconte · 2 years, 6 months ago
- 45e15e3 Prepare for migrating to new AddPeer method by Niels Möller · 2 years, 6 months ago
- 646a907 [DVQA] Remove old constructor from StatsKey by Artem Titov · 2 years, 6 months ago
- 6d05797 StatsEndToEndTests: Remove unneeded fake renderer in tests. by Asa Persson · 2 years, 6 months ago
- f47a724 New struct PeerNetworkDependencies by Niels Möller · 2 years, 6 months ago
- 89f1aa5 Roll chromium_revision 0755db8caa..866506397b (944150:944390) by chromium-webrtc-autoroll · 2 years, 6 months ago
- f05f282 Allow setting volume on unstarted AudioRtpReceivers by Tony Herre · 2 years, 6 months ago
- a9730e5 Roll chromium_revision 2b49f0a621..0755db8caa (944047:944150) by chromium-webrtc-autoroll · 2 years, 6 months ago
- 8ef7da7 [DVQA] Remove sender from DVQA StatsKey by Artem Titov · 2 years, 6 months ago
- f002e2f Roll chromium_revision 40bb827aa5..2b49f0a621 (943489:944047) by chromium-webrtc-autoroll · 2 years, 6 months ago
- 520b365 Fix WebRTC version updater bot. by Mirko Bonadei · 2 years, 6 months ago
- 984cf9b Explicitly set encoder and decoder format in codec tests. by Sergey Silkin · 2 years, 6 months ago
- 8498b7e Revert "Add dav1d decoder to WebRTC." by Artem Titov · 2 years, 6 months ago
- 1478585 Add dav1d decoder to WebRTC. by philipel · 2 years, 6 months ago
- ae47cf7 Optimize suppression_filter. by cschuldt · 2 years, 6 months ago
- ad3551a Roll chromium_revision 20e23fa31a..40bb827aa5 (943065:943489) by Mirko Bonadei · 2 years, 6 months ago
- 524a422 Modify global variables to not call the destructor on exit. by Byoungchan Lee · 2 years, 6 months ago
- 9c4e89d Fixed CFBundleVersion version in Python 3. by Yura Yaroshevich · 2 years, 6 months ago
- 68f06af WebRtcVideoChannelBaseTest.InvalidRecvBufferSize: fix UAF. by Markus Handell · 2 years, 6 months ago
- 4be518d Update autoroll script to support py3 (only) by Christoffer Jansson · 2 years, 6 months ago
- d7eef66 VideoStreamEncoder: move PostTasks to WebRtcVideoChannel. by Markus Handell · 2 years, 6 months ago
- 20d0323 Dump json output for webrtc tests using gtest_parallel. by Jeremy Leconte · 2 years, 6 months ago
- dc3a4c4 Dump json output for webrtc tests using gtest_parallel. by Jeremy Leconte · 2 years, 6 months ago
- a83f874 AGC2 limiter: faster recovery by Alessio Bazzica · 2 years, 6 months ago
- 1a75d5e Remove mentions to xcode_clang which has been removed. by Mirko Bonadei · 2 years, 6 months ago
- 36cbcf1 Remove TBR from version updater script. by Mirko Bonadei · 2 years, 6 months ago
- 6d2be92 Fix mb.py presubmit. by Jeremy Leconte · 2 years, 6 months ago
- 9a478b5 VideoStreamEncoder: expect frame entry on the encoder queue. by Markus Handell · 2 years, 6 months ago
- 7ec990d Fix flakky test: ensure that DVQA was executed for some time. by Artem Titov · 2 years, 6 months ago
- 50b9552 Reintroduce enable_dtls_srtp option by Harald Alvestrand · 2 years, 6 months ago
- 1dfce5d Roll chromium_revision ca9862fcfe..20e23fa31a (942948:943065) by chromium-webrtc-autoroll · 2 years, 6 months ago
- cabc3e5 Delete obsolete method QueryVideoEncoder by Niels Möller · 2 years, 6 months ago
- 0a7a8e0 Remove DEPRECATED_AsyncInvoker::Flush. by Tommi · 2 years, 6 months ago
- 6002b15 Optimize ComputeFrequencyResponse(). by cschuldt · 2 years, 6 months ago
- 7500552 Fix WebRTC version updater bot. by Mirko Bonadei · 2 years, 6 months ago
- 7b0c6a3 Roll chromium_revision ed9be3a428..ca9862fcfe (942828:942948) by chromium-webrtc-autoroll · 2 years, 6 months ago
- be1a170 Roll chromium_revision 35d97f39f3..ed9be3a428 (942706:942828) by chromium-webrtc-autoroll · 2 years, 6 months ago
- 9e677f0 Roll chromium_revision f8776f0d78..35d97f39f3 (942535:942706) by chromium-webrtc-autoroll · 2 years, 6 months ago
- 4de9944 Delete memory allocated by GetStreamCaps by Sergey Silkin · 2 years, 7 months ago
- 9ebe6d7 Remove the AsyncInvoker alias. by Tommi · 2 years, 6 months ago
- 9456501 Try to avoid to set Code-Review+1 by Mirko Bonadei · 2 years, 6 months ago
- 96ef2bd Roll chromium_revision f1aab63b8b..f8776f0d78 (942294:942535) by chromium-webrtc-autoroll · 2 years, 6 months ago
- be9c40f Fix documentation for VideoQualityAnalyzerInterface::GetStreamLabel by Artem Titov · 2 years, 6 months ago
- 9c14573 Add method to return frames history from DVQA. by Artem Titov · 2 years, 6 months ago
- 5a9be30 Remove self TBR to avoid setting Code-Review+1. by Mirko Bonadei · 2 years, 6 months ago
- 2ae54b9 Make Chromium DEPS autoroller set Bot-Commit+1 by Mirko Bonadei · 2 years, 6 months ago
- cc7db01 Roll chromium_revision b56d8868f3..f1aab63b8b (941610:942294) by chromium-webrtc-autoroll · 2 years, 6 months ago
- 607513e Add more usrsctp TSAN suppressions by Florent Castelli · 2 years, 6 months ago
- d823259 Set the maximum number of audio channels to 24 by Ivo Creusen · 2 years, 6 months ago
- 584e3f9 stats collector test: remove is_remote expectations by Philipp Hancke · 2 years, 6 months ago
- f94c053 VSS: Fix TSAN error related to internal variables by Florent Castelli · 2 years, 6 months ago
- 3395107 Remove old definition of the macro RTC_NOTREACHED by Artem Titov · 2 years, 6 months ago
- c09b14c Use a FQDN hostname as an invalid hostname by Florent Castelli · 2 years, 6 months ago
- 4cbfe41 Fix out-of-bounds memory access due to large number of audio channels. by Ivo Creusen · 2 years, 6 months ago
- 13d1636 Delete support for has_internal_source by Niels Möller · 2 years, 6 months ago
- efe46b6 Change the type of RTCVideoSourceStats.framesPerSecond by Byoungchan Lee · 2 years, 7 months ago
- 45448e9 Remove slow_tests by Evan Shrubsole · 2 years, 6 months ago
- 83ceb90 Fix -Wunused-but-set-variable by Byoungchan Lee · 2 years, 6 months ago
- f0f4743 Fix RepeatingTask unittest flakiness by Evan Shrubsole · 2 years, 6 months ago
- d325196 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED by Artem Titov · 2 years, 6 months ago
- f5ea3b9 Roll chromium_revision 867b87eb29..b56d8868f3 (933713:941610) by Mirko Bonadei · 2 years, 6 months ago
- 21c4b1e stats: expose relayProtocol on prflx candidate by Philipp Hancke · 2 years, 7 months ago
- 8718f58 Correctly set first/last packet of frame bit in VideoRtpDepacketizerVp9. by philipel · 2 years, 6 months ago
- 60c01cc setCodecPreferences: dont insert red codec twice by Philipp Hancke · 2 years, 6 months ago
- 62bb58f sdp: check for token-char in C++ style by Philipp Hancke · 2 years, 6 months ago
- 842c209 Remove unused dependency in pacing module by Jianhui Dai · 2 years, 6 months ago
- b8f7ef1 Permit current queue reference to be null on sequence checker creation by Harald Alvestrand · 2 years, 6 months ago
- 5fb1190 Remove infra/tools/luci/isolated from DEPS. by Mirko Bonadei · 2 years, 6 months ago
- 528e489 Set correct spatial layer number in FrameEncodeMetadataWriter by Zhaoliang Ma · 2 years, 7 months ago
- 069539e StatsEndToEndTest.VerifyNackStats: Fix flaky test. by Asa Persson · 2 years, 7 months ago
- bf08745 Implement RTCOutboundRtpStreamStats.targetBitrate for audio. by Jakob Ivarsson · 2 years, 7 months ago
- 1d73243 Use a new instance of RTP stack for each test. by Tim Na · 2 years, 7 months ago
- c86e1c2 StatsEndToEndTest.TestReceivedRtpPacketStats: Fix flaky test. by Asa Persson · 2 years, 7 months ago
- c694270 Update commentary of PacingController by Jianhui Dai · 2 years, 7 months ago
- 187e9d4 sdp: limit mid length to 16 bytes by Philipp Hancke · 2 years, 7 months ago
- fbd52c0 fix stats collector unit test by Philipp Hancke · 2 years, 7 months ago
- 0072c21 Use unique_ptr in GetNextFrame instead of release/delete by Evan Shrubsole · 2 years, 7 months ago
