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885c775
SCReAMv2: Set default NumberOfRttsBetweenResetRefWindowIOnCongestion to 100000000
by Per K
· 68 minutes ago
main
master
96cd279
Remove the deprecated CallConfig worker queue
by Tommi
· 4 hours ago
lkgr
22e5228
sdp: accept TCP/DTLS/RTP/SAVPF proto name
by Philipp Hancke
· 4 hours ago
5098a7e
pc: Clean up worker_thread in RTP senders and receivers
by Tommi
· 4 hours ago
8f3cbff
Make rtc_test private and update the documentation.
by Jeremy Leconte
· 6 hours ago
4139273
Implement OnFrameDropped() for ScalabilityStructureSimulcast.
by Erik Språng
· 7 hours ago
8bb5e8c
datachannel: let a send run pending OnTransportReady for announced channels
by Philipp Hancke
· 7 hours ago
2fb5893
Enforce non-current thread check in Thread::Join
by Tomas Gunnarsson
· 7 hours ago
dff039b
Convert the last rtc_test targets to rtc_test_suite.
by Jeremy Leconte
· 9 hours ago
0732b80
Fire OnSignalingChange after applying the media descriptions
by Jonas Oreland
· 9 hours ago
1792db3
Gather stats in one task on the network thread
by Tommi
· 9 hours ago
84b1496
Handle PeerConnection release in GetStats callbacks
by Tomas Gunnarsson
· 9 hours ago
5d71bc1
Update SCReAMv2 ReduceJitter to latest draft
by Per K
· 11 hours ago
ac03397
Update WebRTC code version (2026-11-06T04:06:52).
by webrtc-version-updater
· 13 hours ago
92a030d
Add encoding_complexity to RtpEncodingParameters
by Erik Språng
· 25 hours ago
c111ae6
Handle PC and factory release from callbacks
by Tommi
· 26 hours ago
80e3ea7
Remove a few unused variables
by Nico Weber
· 26 hours ago
827424a
Split Android swcodecs_jni into decoder and encoder targets
by Danil Chapovalov
· 27 hours ago
09da4a3
Record the outcome of the WGC window capturer fallback
by Palak Agarwal
· 27 hours ago
e34fdcb
Implement OnFrameDropped() and add unit tests for ScalableVideoControllerNoLayering.
by Erik Språng
· 28 hours ago
c5d98f7
Implement OnFrameDropped() for ScalabilityStructureKeySvc.
by Erik Språng
· 28 hours ago
9299c83
Remove unused reuse_streams option from video timing simulator.
by Rasmus Brandt
· 28 hours ago
1633e94
Remove RTC_LOG calls from Android JNI_OnLoad
by Tommi
· 30 hours ago
6189ba7
Make frame transformer short circuiting reversible
by Tommi
· 31 hours ago
05b796d
Saturate output of WebRtcSpl_AffineTransformVector
by karllen.zheng
· 31 hours ago
29e428e
[vpython] Upgrade brotli from 1.0.9 to 1.2.1
by Alex Ovsienko
· 33 hours ago
9fcbd99
Convert rtc_test targets that don't run on CI to rtc_executable.
by Jeremy Leconte
· 33 hours ago
cf6a5da
Cleanup expired unused field trial WebRTC-DscpFieldTrial
by Danil Chapovalov
· 34 hours ago
b2fbca1
Add rtc_xctest for iOS XCTest bundles.
by Jeremy Leconte
· 35 hours ago
5e07d59
Roll chromium_revision e625867486..3a516cc8e0 (1735542:1735986)
by chromium-webrtc-autoroll
· 2 days ago
8ee4e84
Update WebRTC code version (2026-11-05T04:06:48).
by webrtc-version-updater
· 2 days ago
3075ede
Roll chromium_revision f1f63e7d67..e625867486 (1735212:1735542)
by chromium-webrtc-autoroll
· 2 days ago
838f95e
Roll chromium_revision acccc247a1..f1f63e7d67 (1734313:1735212)
by chromium-webrtc-autoroll
· 2 days ago
5c003a2
Link with an implementation for video_capture_module in new unittest
by Danil Chapovalov
· 2 days ago
1f18df5
Remove BaseChannel::worker_thread and parameter
by Tommi
· 2 days ago
c00a1ce
pc: Remove ConnectionContext::worker_thread() and worker_thread_
by Tommi
· 2 days ago
1699a0d
Call: Unify worker and network threads in Call and CallConfig
by Tommi
· 2 days ago
3869ec1
Migrate AudioSendStream and VideoSendStreamImpl to network queue
by Tommi
· 2 days ago
b85d54e
Increase pacer delay expectation in ScreamTest.LinkCapacity2MbitRepeatedDelaySpikesNoEcn
by Per Kjellander
· 2 days ago
7001dbb
Run camera-independent video capture tests on CI
by Jeremy Leconte
· 2 days ago
ba72751
Implement OnFrameDropped() for ScalabilityStructureFullSvc.
by Erik Språng
· 2 days ago
c87278f
Expose BuildSframeDescriptorByte()
by kwasniow
· 2 days ago
08dca3a
Pass the Sframe encryption config to audio send streams
by kwasniow
· 2 days ago
928c7e9
Make target_delay_unittest and read_auth_file_unittest rtc_cc_test.
by Jeremy Leconte
· 2 days ago
e45c05e
p2p: ignore gathering done of generation replaced by an ICE restart
by Philipp Hancke
· 2 days ago
2cf0adb
Update WebRTC code version (2026-11-04T04:05:31).
by webrtc-version-updater
· 3 days ago
457226f
Reject enabling Sframe without a media channel
by kwasniow
· 3 days ago
a057d00
Roll chromium_revision 1cb559f887..acccc247a1 (1734022:1734313)
by chromium-webrtc-autoroll
· 3 days ago
3bdd69e
Require a non-empty Sframe decrypt error callback
by kwasniow
· 3 days ago
502349e
Roll chromium_revision 4c6d6a8665..1cb559f887 (1733832:1734022)
by chromium-webrtc-autoroll
· 3 days ago
70f274a
Add ScalableVideoController::OnFrameDropped().
by Erik Språng
· 3 days ago
19581c3
Reject SDP audio media descriptions where maxptime is less than minptime.
by Henrik Boström
· 3 days ago
36c0dd4
Move goog_cc_test under peer_scenario/bwe_integration_tests
by Jeremy Leconte
· 3 days ago
0ea89ec
Roll chromium_revision fbe9dac83e..4c6d6a8665 (1733699:1733832)
by chromium-webrtc-autoroll
· 3 days ago
da583a1
Migrate VideoReceiveStream2 and RtpVideoStreamReceiver2 to network queue
by Tommi
· 3 days ago
685ec51
Pass the webrtc-changelog summary via a file
by Jeremy Leconte
· 3 days ago
f3c1a4c
video_capture: Reject unusable PipeWire frame chunks
by Viorel Cernateanu
· 3 days ago
58f447a
Migrate AudioReceiveStream and ChannelReceive to network thread
by Tommi
· 3 days ago
3c8b937
Re-enable FrameSampler unit tests
by Jeremy Leconte
· 3 days ago
84cb55b
Roll chromium_revision f34bff24b9..fbe9dac83e (1733571:1733699)
by chromium-webrtc-autoroll
· 3 days ago
802e857
Update WebRTC code version (2026-11-03T04:07:49).
by webrtc-version-updater
· 4 days ago
bf7c9f1c
Roll chromium_revision c564c3277f..f34bff24b9 (1732912:1733571)
by chromium-webrtc-autoroll
· 4 days ago
70d65f3
Don't build ScopedLoggingConfig in Chromium
by Tommi
· 4 days ago
a0134ba
Fix null deref in TurnPort::SendTo for addresses without a TurnEntry
by Jonas Oreland
· 4 days ago
5280272
Roll chromium_revision 7193e93838..c564c3277f (1732661:1732912)
by chromium-webrtc-autoroll
· 4 days ago
ede1bc2
Limit configured STUN servers to 32
by Tommi
· 4 days ago
8c1451f
Deprecate legacy LogMessage configuration methods
by Tommi
· 4 days ago
74fce7a
Add ScopedLoggingConfig test helper
by Tommi
· 4 days ago
6b54e7c
Remove unneeded apm() and engine() getters in WebRtcVoiceEngine
by Sam Zackrisson
· 4 days ago
368aa4f
Add congestion control feedback report wire format in RtcTransport
by Kaustav Ghosh
· 4 days ago
10a4b2f
Migrate tests away from LogMessage::AddLogToStream
by Tommi
· 4 days ago
0b6fb16
Neural residual echo estimator: initialize with normal thread prio
by Sam Zackrisson
· 4 days ago
d677f6c
fuzzer: update SDP fuzzer
by Philipp Hancke
· 4 days ago
483381d
Roll chromium_revision df60081729..7193e93838 (1732505:1732661)
by chromium-webrtc-autoroll
· 4 days ago
8290a27
Trim redundant phases from Windows more_configs
by Fredrik Solenberg
· 4 days ago
11bee9b
Remove obsolete reproxy.cfg generation from DEPS
by Fumitoshi Ukai
· 4 days ago
681074e
Update WebRTC code version (2026-11-02T04:05:42).
by webrtc-version-updater
· 5 days ago
8838a7c
Roll chromium_revision 75203ddc49..df60081729 (1732398:1732505)
by chromium-webrtc-autoroll
· 5 days ago
556f7ee
Roll chromium_revision 8d06155339..75203ddc49 (1732284:1732398)
by chromium-webrtc-autoroll
· 5 days ago
13989ae
Update WebRTC code version (2026-11-01T04:06:11).
by webrtc-version-updater
· 6 days ago
977d873
Roll chromium_revision b2f4726af5..8d06155339 (1732180:1732284)
by chromium-webrtc-autoroll
· 6 days ago
da453ef
Roll chromium_revision a9757fc63f..b2f4726af5 (1732067:1732180)
by chromium-webrtc-autoroll
· 6 days ago
5ac2dde
Update WebRTC code version (2026-10-31T04:09:30).
by webrtc-version-updater
· 7 days ago
031cc21
Roll chromium_revision 0033b2d8a8..a9757fc63f (1731780:1732067)
by chromium-webrtc-autoroll
· 7 days ago
ba43b08
Remove loss notification (goog-lntf) support
by Tommi
· 7 days ago
ffa1c03
Roll chromium_revision 3dab22bdef..0033b2d8a8 (1731663:1731780)
by chromium-webrtc-autoroll
· 7 days ago
be71c12
Roll chromium_revision dcd9b11b9f..3dab22bdef (1731338:1731663)
by chromium-webrtc-autoroll
· 7 days ago
a8618a7
Roll chromium_revision af530c1a32..dcd9b11b9f (1730047:1731338)
by chromium-webrtc-autoroll
· 7 days ago
5a07fa4
cryptex: change UseCryptex to take the policy and negotiation result
by Philipp Hancke
· 7 days ago
bf496c4
Deflake unsignaled receiver getStats integration tests
by Tommi
· 7 days ago
8483eaa
Remove GetLoggingLock()
by Tommi
· 7 days ago
f7738c0
Reset existing receive frame transformer delegate
by Tommi
· 7 days ago
269c658
Ignore repeated frames in LossNotificationController
by Tommi
· 7 days ago
d99a70b
Migrate A/V stream synchronization to network thread
by Tommi
· 7 days ago
75ca5bd
Run CQ tests in release and compile debug, like Chromium
by Fredrik Solenberg
· 7 days ago
2a5e358
Roll chromium_revision a0c7fb6108..af530c1a32 (1728225:1730047)
by Christoffer Dewerin
· 7 days ago
f7df80a
Restrict peerconnection example to loopback.
by Henrik Boström
· 7 days ago
6ae29a6
Parse DEPS files with gclient_eval in roll_deps.py
by Jeremy Leconte
· 7 days ago
8d56a08
Move legacy log sinks off the global logging lock
by Tommi
· 7 days ago
3178a28
Update WebRTC code version (2026-10-30T04:11:07).
by webrtc-version-updater
· 8 days ago
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