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d9bd07b
Roll chromium_revision 464ebda2ac..779c058eac (1696311:1696434)
by chromium-webrtc-autoroll
· 31 minutes ago
main
master
06acbbc
Roll chromium_revision 85078ad0ec..464ebda2ac (1696126:1696311)
by chromium-webrtc-autoroll
· 2 hours ago
9fc6fae
Add screenshare CBR rate control tests for V2 Video Encoder API.
by Erik Språng
· 6 hours ago
lkgr
ea28a89
Delete WebRTC-Target-Bitrate-Rtcp field trial
by Danil Chapovalov
· 6 hours ago
5adbd93
Roll chromium_revision 5334a28253..85078ad0ec (1696014:1696126)
by chromium-webrtc-autoroll
· 6 hours ago
c2099d1
Embed rtc_rust check into rtc_rust_test_suite template
by Danil Chapovalov
· 7 hours ago
22dfd42
scale_resolution_down_to: Ensure lowest active stream is not disabled.
by Åsa Persson
· 8 hours ago
e4edcbb
Fix building frame_analyzer_host target
by Danil Chapovalov
· 9 hours ago
8244888
Roll chromium_revision db0bd2e3b9..5334a28253 (1695900:1696014)
by chromium-webrtc-autoroll
· 10 hours ago
9797914
Update WebRTC code version (2026-09-11T04:11:18).
by webrtc-version-updater
· 14 hours ago
d077e12
Roll chromium_revision 032c48a8ba..db0bd2e3b9 (1695697:1695900)
by chromium-webrtc-autoroll
· 16 hours ago
3c93c04
Roll chromium_revision af86ccbd91..032c48a8ba (1695463:1695697)
by chromium-webrtc-autoroll
· 21 hours ago
35b68b1
Roll chromium_revision de383d292c..af86ccbd91 (1695310:1695463)
by chromium-webrtc-autoroll
· 27 hours ago
04408f0
Document current state of Rust support in WebRTC
by Danil Chapovalov
· 27 hours ago
782a8de
Remove rust libraries matching cxx bridges
by Danil Chapovalov
· 27 hours ago
68c958e
Extract common Y4M header parser and add unit tests
by Erik Språng
· 30 hours ago
dc39512
Add high complexity CBR test cases for V2 Video Encoder API.
by Erik Språng
· 30 hours ago
e101a4c
Invoke Audio Injector callback on ProxyAudioEncoder::OnReceivedTargetAudioBitrate calls.
by Leonardo Evi
· 30 hours ago
84155da
opus: remove unused WebRtcOpus_PacketHasVoiceActivity
by Philipp Hancke
· 30 hours ago
b40276d
Fix RtpSenderReceiverTest.InjectVideoFrameOnDifferentThread flakey test.
by Leonardo Evi
· 30 hours ago
0999bd5
Roll chromium_revision 34204b996b..de383d292c (1695158:1695310)
by chromium-webrtc-autoroll
· 32 hours ago
f9e1d92
Deflake ReceivedUlpfecPacketsNotNacked test
by Tommi
· 32 hours ago
7de2c30
Batch transceiver channel teardown in ApplyRemoteDescription
by Tommi
· 32 hours ago
bf31ff6
pc: check remote ice-option validity and split ufrag/pwd UMA
by Philipp Hancke
· 34 hours ago
c61fc59
Default worker thread to network thread
by Tommi
· 35 hours ago
9a6c973
Update WebRTC code version (2026-09-10T04:10:51).
by webrtc-version-updater
· 2 days ago
3c3e048
Roll chromium_revision 7b8e3afa63..34204b996b (1694827:1695158)
by chromium-webrtc-autoroll
· 2 days ago
ee390f4
Roll chromium_revision 7a79a07b9c..7b8e3afa63 (1694328:1694827)
by chromium-webrtc-autoroll
· 2 days ago
c274fd8
Clean up dynamic configuration of logging in tests and default to LS_WARNING level logging.
by Fredrik Solenberg
· 2 days ago
dab572f
Fix flaky audio delay test by tracking samples cumulatively
by Tommi
· 2 days ago
021f96f
Fix a bug where ICE restart with the same address would disconnect the selected connection on the callee side.
by Kun Wu
· 2 days ago
348e5ff
Revert "RtpSender: Create the Sframe encryptor and hand out a worker proxy"
by Ilya Nikolaevskiy
· 2 days ago
0520f29
Update protobuf version to fix perf bots
by Jeremy Leconte
· 2 days ago
8354b7c
RtpSender: Create the Sframe encryptor and hand out a worker proxy
by kwasniow
· 2 days ago
1e381bc
Reenable AudioPacketLoss test
by Danil Chapovalov
· 2 days ago
0605df5
dtls-in-stun: remove invalid dashes from ice-option
by Philipp Hancke
· 2 days ago
21c61a7
Validate NALU length and SPS dimensions in H.264 RTP depacketizer
by Erik Språng
· 2 days ago
1ebdcdf
Replace ArrayView with std::span in pc/packet_processor
by Danil Chapovalov
· 3 days ago
c7add88
Update WebRTC code version (2026-09-09T04:04:24).
by webrtc-version-updater
· 3 days ago
2d69cb6
Roll chromium_revision 9ec5bfbaba..7a79a07b9c (1694198:1694328)
by chromium-webrtc-autoroll
· 3 days ago
0d64c7d
Roll chromium_revision 89afa034bb..9ec5bfbaba (1694098:1694198)
by chromium-webrtc-autoroll
· 3 days ago
55839bc
Roll chromium_revision 1a5eaa35af..89afa034bb (1693924:1694098)
by chromium-webrtc-autoroll
· 3 days ago
683588c
Roll chromium_revision 0739e1d2d9..1a5eaa35af (1693673:1693924)
by chromium-webrtc-autoroll
· 3 days ago
d4443ab
Remove legacy logging calls
by Fredrik Solenberg
· 3 days ago
c39a2fc
Add simulated time support to CallTest
by Tomas Gunnarsson
· 3 days ago
08a639c
Protect PeerConnection lifecycle and JNI invocations with ReadWriteLock.
by Linus Nilsson
· 3 days ago
8a9a929
Prevent delayed FrameDecoded from advancing RTP cleanup epoch
by roman.yuanm
· 3 days ago
05e27f1
Propagate DTLS-SRTP writable state changes after activation
by Shumin Xiao
· 3 days ago
88fae9d
Roll chromium_revision 89fd0aa10a..0739e1d2d9 (1693561:1693673)
by chromium-webrtc-autoroll
· 3 days ago
b7f122d
Migrate standalone tools and loopback tests to LoggingConfig
by Fredrik Solenberg
· 3 days ago
83c2ab9
pc: fix premature ICE candidate signaling in CallTransferredForCaller
by Tommi
· 3 days ago
52e450c
Add CBR rate control tests for V2 VideoEncoderInterface validation.
by Erik Språng
· 3 days ago
23b0662
Fix race in TooManyDataChannelsOpenedBeforeConnecting
by Tommi
· 4 days ago
47142fa
Roll chromium_revision 12cc78a34e..89fd0aa10a (1693454:1693561)
by chromium-webrtc-autoroll
· 4 days ago
0edddd4
Update WebRTC code version (2026-09-08T04:11:54).
by webrtc-version-updater
· 4 days ago
926a9e7
Roll chromium_revision ff37f76274..12cc78a34e (1693276:1693454)
by chromium-webrtc-autoroll
· 4 days ago
7d8c903
Add OnSourceChanged to RtpReceiverObserverInterface
by Tommi
· 4 days ago
175760c
Eliminate unnecessary copies in common_audio/
by Fredrik Solenberg
· 4 days ago
146abab
Increase execution timeout for webrtc_builders to 3 hours
by Christoffer Dewerin
· 4 days ago
99e7859
Propagate RtpPacketInfos to audio OnData sink callbacks.
by Tommi
· 4 days ago
e709ce2
Validate dimensions and crop offsets in H.264 SPS parser
by Erik Språng
· 4 days ago
87e745f
Filter ambiguous receive times and make delay metrics optional in SCReAM
by Per K
· 4 days ago
a6fd926
Roll chromium_revision dca9c4468e..ff37f76274 (1690475:1693276)
by chromium-webrtc-autoroll
· 4 days ago
5e94740
pc: resolve chained AddIceCandidate calls in order
by Philipp Hancke
· 4 days ago
61b894d
Delete ConnectionContext::use_rtx
by Danil Chapovalov
· 4 days ago
21a9800
pc: use ssrc_s() in stats collectors on signaling thread
by Tommi
· 4 days ago
a722beb
Add cl-description skill for authoring CL messages
by Tommi
· 4 days ago
e9fc3ee
pc: trace restartIce and addIceCandidate with result/error
by Philipp Hancke
· 4 days ago
2284898
pc: trace create, addTrack and addTransceiver
by Philipp Hancke
· 4 days ago
29be127
Add audio_level, absolute_capture_timestamp and csrcs overrides to audio injection path.
by Leonardo Evi
· 4 days ago
f72faef
Add SwitchingFrameReader test utility.
by Erik Språng
· 4 days ago
5fd97c1
Roll chromium_revision 57ed38e9f7..dca9c4468e (1682199:1690475)
by Jeremy Leconte
· 4 days ago
9bc0124
Introduce EncodedAudioFrameInjectorInterface.
by Leonardo Evi
· 4 days ago
5155d4b
Decouple transport packet delivery from network thread in CallTest
by Tomas Gunnarsson
· 4 days ago
6efdb1d
Update WebRTC code version (2026-09-07T04:05:43).
by webrtc-version-updater
· 5 days ago
a15ee91
Update WebRTC code version (2026-09-06T04:07:53).
by webrtc-version-updater
· 6 days ago
0938548
SCReAM: Only calculate delay metrics on packets in recent burst window
by Per K
· 6 days ago
fee771a
Add SocketReceiveTimeCalculator
by Per K
· 6 days ago
32618d8
Update WebRTC code version (2026-09-05T04:09:05).
by webrtc-version-updater
· 7 days ago
247e438
Remove sending rtcp TargetBitrate for screencast by default
by Danil Chapovalov
· 7 days ago
c4f5b6e
pc: trace OnTrack events
by Philipp Hancke
· 7 days ago
623cf7b
Add VideoEncoderRateControlTest for v2 VideoEncoderInterface.
by Erik Språng
· 7 days ago
6bc6a2f
Remove data_deps from audioproc_test_utils.
by Jeremy Leconte
· 7 days ago
42c9bca
Avoid some unnecessary copying in video_frame_buffer
by Fredrik Solenberg
· 7 days ago
6ba4f10
Introduce EncodedVideoFrameInjectorInterface.
by Leonardo Evi
· 7 days ago
8a63c9a
Update WebRTC code version (2026-09-04T04:09:59).
by webrtc-version-updater
· 8 days ago
a192095
Clean up LibaomAv1EncoderV2 implementation.
by Erik Språng
· 8 days ago
9ab2b29
Add DTLS wire format framing stub to RtcTransport
by Kaustav Ghosh
· 8 days ago
79edcdc
Fix some audio_processing dependencies
by Jeremy Leconte
· 8 days ago
90e0687
Register sframe license file
by Björn Terelius
· 8 days ago
e0b3658
Cleanup field trial WebRTC-Bwe-ReceiveTimeFix
by Per K
· 8 days ago
8668c11
Add width and height to CreateOutgoingVideoFrame.
by Leonardo Evi
· 8 days ago
42cb1d2
Track audio playout time in ChannelReceive
by Tommi
· 8 days ago
1f2dc30
Avoid some unnecessary copying in video and frame pipeline
by Fredrik Solenberg
· 8 days ago
95773cc
Migrate audio_processing_unittests to be a rtc_test_suite
by Jeremy Leconte
· 8 days ago
4287e09
Clamp arrival_time_offset in TransportFeedbackAdapter
by Per K
· 9 days ago
2499eef
Prefer std::make_tuple over testing::make_tuple
by Philipp Hancke
· 9 days ago
4e2f2ca
Update WebRTC code version (2026-09-03T04:09:34).
by webrtc-version-updater
· 9 days ago
573c5c6
Fix issue where codec fallback skips first codec in list.
by Henrik Boström
· 9 days ago
4046761
Fix RTCConfiguration equality for SCTP SNAP
by Joachim Reiersen
· 9 days ago
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