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d233584
Update WebRTC code version (2026-09-23T04:10:16).
by webrtc-version-updater
· 17 minutes ago
main
master
f08ed17
dtls-in-stun: Disable testcases that can no longer occur
by Jonas Oreland
· 75 minutes ago
d275cd1
Protect Android TurnCustomizer and CallSessionFileRotatingLogSink against double-free.
by Linus Nilsson
· 10 hours ago
lkgr
ec71aad
Add CreatePeerConnectionFactory overload without worker_thread
by Tommi
· 13 hours ago
1852f89
Replace absl attributes with there modern C++ equivalents
by Danil Chapovalov
· 15 hours ago
6808fa8
Prepare VideoSendStreamImpl for asynchronous Stop
by Markus Handell
· 15 hours ago
b8586e0
Set Current() in SimulatedTaskQueue::Delete
by Markus Handell
· 15 hours ago
a39fef9
Remove VideoStreamEncoder::GetAdaptationResources
by Markus Handell
· 15 hours ago
55bdfd7
Add audio_coding_unittests and rtp_rtcp_unittests test suites.
by Jeremy Leconte
· 15 hours ago
94e682e
Roll chromium_revision df36763e68..971f8da90a (1697354:1701880)
by Johannes Kron
· 17 hours ago
dfb97f0
Avoid port collisions in AppRTC Robolectric JUnit tests
by Johannes Kron
· 17 hours ago
7a951d8
Allocate RejectsHugeSize payload on the heap.
by Jeremy Leconte
· 17 hours ago
1e70146
Protect Android DtmfSender and TimestampAligner against UAF.
by Linus Nilsson
· 17 hours ago
8bea72f
Add init readiness metric for ML residual echo estimation
by Sam Zackrisson
· 17 hours ago
57b98b1
sdk: Stop using a distinct worker thread
by Tommi
· 17 hours ago
a98a2e8
Align AEC flag handling with other APM config flags
by Per Åhgren
· 17 hours ago
8e46acc
Add initialization outcome metric for ML residual echo estimation
by Sam Zackrisson
· 18 hours ago
bbc2085
Inject test adaptation resources at construction
by Markus Handell
· 18 hours ago
63f7a75
Set export_linear_aec_output on multichannel AEC config
by Per Åhgren
· 18 hours ago
95fceff
Detach senders from their send stream on rollback
by Tommi
· 18 hours ago
aa0877a
Per temporal layer rate control in the V2 libaom AV1 encoder.
by Erik Språng
· 19 hours ago
f24ae71
Add model load latency metric for ML residual echo estimation
by Sam Zackrisson
· 19 hours ago
767dd4d
dtls-in-stun: skip sped/pqc tests when not built with BoringSSL
by Philipp Hancke
· 20 hours ago
a3553f1
[dcsctp] Immediately send outstanding data after handover
by Priya Rajarathinam
· 22 hours ago
acf6607
Stop sending ECT(1) if the path drops the marking
by Per K
· 22 hours ago
fe36fc7ca
Update WebRTC code version (2026-09-22T04:09:49).
by webrtc-version-updater
· 24 hours ago
48de432
Remove VideoStreamEncoder::encoder_queue()
by Markus Handell
· 2 days ago
7994089
Remove event waits from VideoStreamEncoder tests
by Markus Handell
· 2 days ago
7e4a6a2
Remove restrictions listener test hooks from VideoStreamEncoder
by Markus Handell
· 2 days ago
a9c6ddc
Preserve builder APM and AEC configurations in AudioprocFloat
by Per Åhgren
· 2 days ago
23df7ef
Removed the unused WebRtcSpl_SqrtOfOneMinusXSquared functionality
by Per Ahgren
· 2 days ago
09297de
The ComplexFft functionality beneath common_audio is no longer used.
by Per Ahgren
· 2 days ago
d943abd
Removing bitexactness verification functionality in audioproc_f
by Per Åhgren
· 2 days ago
f4348b8
Protect Android MediaStream, MediaStreamTrack, and MediaSource against UAF.
by Linus Nilsson
· 2 days ago
7b28688
Protect RtpSender, RtpReceiver, and RtpTransceiver against UAF.
by Linus Nilsson
· 2 days ago
0f3da55
Adopt NativeLifecycleLock in DataChannel.java
by Linus Nilsson
· 2 days ago
d553533
Keep send parameters that were not applied
by Tommi
· 2 days ago
98de979
Let encoders publish a max frame size and clamp the source to it
by Andreas Pehrson
· 2 days ago
c0dd1f8
Split FakeVideoRenderer into own helper target
by Danil Chapovalov
· 2 days ago
49fe6a3
Add test for munged local offer with fewer layers
by Tommi
· 2 days ago
5516497
Cleanup p2p unit test + SDP munging detector
by Philipp Hancke
· 2 days ago
6583762
pc: make always_negotiate_datachannels immutable
by Philipp Hancke
· 2 days ago
d40ac5b
dav1d: Handle dav1d_data_wrap failures
by Philipp Hancke
· 2 days ago
27657fd
Skip BasicOfferAnswerPayloadTypesStable if rtc_use_h264=false.
by Mirko Bonadei
· 2 days ago
3a56730
Move the ECT(1) marking decision into Ect1Policy
by Per K
· 2 days ago
d6d8d97
Link Rust code into prod library.
by Danil Chapovalov
· 2 days ago
f98760a
Split ConfigureForRtcpFeedback into two setters
by Per K
· 2 days ago
edbe411
Derive ECT(1) bleaching from PacketResult
by Per K
· 2 days ago
176602d
Keep operations pending through result delivery
by Michael Reeves
· 2 days ago
fb8ad81
Remove 'WebRTC Chromium FYI Win Builder (dbg)' from LKGR
by Jeremy Leconte
· 2 days ago
438e074
Update WebRTC code version (2026-09-21T04:05:20).
by webrtc-version-updater
· 2 days ago
80a2d43
Add unit tests for ChannelReceiveFrameTransformerDelegate
by Tommi
· 3 days ago
3f798e3
Update WebRTC code version (2026-09-20T04:06:59).
by webrtc-version-updater
· 3 days ago
c250ac7
Reconcile send layer count on the signaling thread
by Tommi
· 4 days ago
e7c0982
Update WebRTC code version (2026-09-19T04:07:48).
by webrtc-version-updater
· 4 days ago
e47a101
p2p: listen to socket close events
by Philipp Hancke
· 5 days ago
2d9b4fc
Make the CBR intra frame budget an explicit encoder setting
by Erik Språng
· 5 days ago
4f41278
Remove the GoogCC/Scream wrapper controller
by Per K
· 5 days ago
fb27a1c
dtls-in-stun: add ice-option for embedding to the SDP
by Philipp Hancke
· 5 days ago
b2a7366
Add RtpPacketizerSframe (SFrame RTP packetizer)
by kwasniow
· 5 days ago
72107b0
[agc2] Make background speaker offset configurable via field trial
by Jesús de Vicente Peña
· 5 days ago
cfb487e
Use FakeRenderer when fuzzing
by Danil Chapovalov
· 5 days ago
2960bf4
Disable windows chromium bots.
by Jeremy Leconte
· 5 days ago
b32e4b3
Reimplement in Rust CorruptionDetection RTP header extension parsing
by Danil Chapovalov
· 6 days ago
7a24158
Don't crash when SDP describes fewer send layers
by Tommi
· 6 days ago
dcfe2ae
[vpython] Migrate legacy .vpython3 to modern vpython.toml (UV)
by Alex Ovsienko
· 6 days ago
6c3c1fe
cryptex consistency fixes
by Philipp Hancke
· 7 days ago
7314d79
Defer codec list consistency check to complete lists
by Tommi
· 7 days ago
f731aab
vp9: memset buffers to zero when enlarging them
by Philipp Hancke
· 7 days ago
fbfb49a
stats: only show media-playout when receiving audio
by Philipp Hancke
· 7 days ago
ad898d4
ssl: use SSL_AD_INTERNAL_ERROR on timeout instead of -1
by Philipp Hancke
· 7 days ago
aee44dc
VideoFrameBuffer: introduce ToI420ForInspection
by Markus Handell
· 7 days ago
3c1c052
Relax quality thresholds in VideoCodecTestAv1.VeryLowBitrate
by Johannes Kron
· 7 days ago
a089f84
Update WebRTC code version (2026-09-16T04:06:56).
by webrtc-version-updater
· 7 days ago
26b31d40
Add Sframe decrypt error types and reporter
by kwasniow
· 8 days ago
e803889
Introduce NativeLifecycleLock and protect PeerConnectionFactory against UAF.
by Linus Nilsson
· 8 days ago
d8c5a46
Remove support of sending RTCP TargetBitrate from the RTP layer
by Danil Chapovalov
· 8 days ago
ae00e08
dav1d: Remove dead code from Dav1dDecoder
by Philipp Hancke
· 8 days ago
74a0ae5
Fix payload type collision when creating a re-offer
by Tommi
· 8 days ago
dcab586
Fix VP9 test flake caused by pacer key frame flush
by Tommi
· 8 days ago
02bd855
Log bitrate values in GoogCcNetworkController
by Tommi
· 8 days ago
b629980
Fix test failures triggered by vector/string allocation changes.
by Mirko Bonadei
· 8 days ago
a2a4043
opus: fix GeneratePlc for stereo and requested length
by Philipp Hancke
· 8 days ago
b7ddef1
Update WebRTC code version (2026-09-15T04:05:07).
by webrtc-version-updater
· 8 days ago
bbd8208
[Wayland] Fix crash in ProcessDMABuffer and clean up renegotiate event
by Jordan Bayles
· 8 days ago
9ae294c
Roll chromium_revision fb0b324870..df36763e68 (1697189:1697354)
by chromium-webrtc-autoroll
· 8 days ago
8d208e7
Remove SetUseNetworkThreadAsWorkerThread
by Tommi
· 9 days ago
96b154a
audio_coding: prefer CHECK*(actual, expected)
by Philipp Hancke
· 9 days ago
4c108c7
Roll chromium_revision 7191362d2c..fb0b324870 (1697086:1697189)
by chromium-webrtc-autoroll
· 9 days ago
d5018f6
doc: recommend --no-history for CI
by Philipp Hancke
· 9 days ago
e78e77b
opus: misc cleanup
by Philipp Hancke
· 9 days ago
3b12ab6
Stop supplying a distinct worker thread in tests
by Tommi
· 9 days ago
559db48
Roll chromium_revision 716ea38d18..7191362d2c (1696985:1697086)
by chromium-webrtc-autoroll
· 9 days ago
6b4978c
Warn when a distinct worker thread is configured
by Tommi
· 9 days ago
a058820
In Video EncoderSink delete video allocation callback
by Danil Chapovalov
· 9 days ago
b9eac31
[agc2] Differentiate target vs speech probability to handle background speakers in the experimental (non active) code branch.
by Jesús de Vicente Peña
· 9 days ago
1079dd2
Update WebRTC code version (2026-09-14T04:06:08).
by webrtc-version-updater
· 9 days ago
3bbfdf3
AVX2: Update AVX2 cpu info based on MSC_VER
by Uday M Bhat
· 9 days ago
e42b7b0
Roll chromium_revision 63eab029b0..716ea38d18 (1696881:1696985)
by chromium-webrtc-autoroll
· 9 days ago
be0e900
Update WebRTC code version (2026-09-13T04:10:34).
by webrtc-version-updater
· 10 days ago
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