Google Git
Sign in
webrtc/src/HEAD
  1. dd6f398 Roll chromium_revision a5481b8c3d..854ba7d446 (1631958:1632086) by chromium-webrtc-autoroll · 57 minutes ago lkgr main master
  2. f9e5ef2 Add RtpDemuxer::RemoveAllSinks and tests. by Tommi · 3 hours ago
  3. c1d536c Refresh style guide by Danil Chapovalov · 4 hours ago
  4. 5b43503 In video_frame.cc, add a null check for the result of VideoFrameBuffer::ToI420() within NativeToJavaVideoFrame to prevent crashes if the conversion fails. by Magnus Jedvert · 5 hours ago
  5. 3450504 Cap RTCConfiguration certificates and enforce 32-bit overflow checks by Tommi · 7 hours ago
  6. 1132973 Update WebRTC code version (2026-05-18T04:09:14). by webrtc-version-updater · 8 hours ago
  7. 197ec92 Roll chromium_revision 5f91be0811..a5481b8c3d (1631846:1631958) by chromium-webrtc-autoroll · 9 hours ago
  8. f172c4f Roll chromium_revision 85a6061cf2..5f91be0811 (1631741:1631846) by chromium-webrtc-autoroll · 31 hours ago
  9. fab9f2e Roll chromium_revision c552e63987..85a6061cf2 (1631051:1631741) by chromium-webrtc-autoroll · 2 days ago
  10. a808a6c Update WebRTC code version (2026-05-16T04:10:33). by webrtc-version-updater · 2 days ago
  11. 7ff59ed Wayland capture: Fix integer overflow in cursor bitmap validation by Jan Grulich · 3 days ago
  12. b343cd4 Validate CGImage dimensions in MouseCursorMonitorMac by Johannes Kron · 3 days ago
  13. 9310b29 Fix race conditions in RTCPeerConnectionFactoryTests by Tommi · 3 days ago
  14. 9e36f71 Synchronize capture queue before verifying mock expectations by Tommi · 3 days ago
  15. 727c9f4 Roll chromium_revision 9e27e6ac5f..c552e63987 (1630693:1631051) by chromium-webrtc-autoroll · 3 days ago
  16. b9f0b18 Guard RestoreTokenManager add/reads with Mutex by Alexander Cooper · 4 days ago
  17. c5bf6f1 Roll chromium_revision d1c637db2e..9e27e6ac5f (1630466:1630693) by chromium-webrtc-autoroll · 4 days ago
  18. 718e2df [PT Redesign] Implement late payload type allocation for video. by Harald Alvestrand · 4 days ago
  19. 264b57c Revert "srtp: add UseCryptex API to SrtpSession and SrtpTransport" by Harald Alvestrand · 4 days ago
  20. b9386f3 srtp: add UseCryptex API to SrtpSession and SrtpTransport by Philipp Hancke · 4 days ago
  21. df2dce4 Improve SrtpSession thread safety and modernize sequence checking by Tommi · 4 days ago
  22. c2286d8 Update WebRTC code version (2026-05-14T04:07:37). by webrtc-version-updater · 4 days ago
  23. e794c6d Roll chromium_revision d8b2c47057..d1c637db2e (1630331:1630466) by chromium-webrtc-autoroll · 4 days ago
  24. 1394980 Roll chromium_revision f82317413d..d8b2c47057 (1629860:1630331) by chromium-webrtc-autoroll · 5 days ago
  25. 5f5e7b4 Minor updates to WebRTC Video Engine by Tommi · 5 days ago
  26. 0ffbc6e Use real rather than simulated task queues in rtp replayer fuzzers by Danil Chapovalov · 5 days ago
  27. ce8b188 Roll chromium_revision 15ef075048..f82317413d (1629755:1629860) by chromium-webrtc-autoroll · 5 days ago
  28. 536cb10 Simplify WebRTC Voice Engine, remove `friend`. by Tommi · 5 days ago
  29. 1cb0465 Remove RtpPacketSinkInterface inheritance from ReceiveStatisticsImpl by Tommi · 5 days ago
  30. 4356919 Update WebRTC code version (2026-05-13T04:05:03). by webrtc-version-updater · 5 days ago
  31. c1903e5 Roll chromium_revision 8af2e3f3e3..15ef075048 (1629518:1629755) by chromium-webrtc-autoroll · 5 days ago
  32. d83af38 Make some VideoReceiveStream2 and RtpVideoStreamReceiver2 members const by Tommi · 6 days ago
  33. 11b5447 Fix UB when comparing two empty webrtc::Buffer objects by Joachim Reiersen · 6 days ago
  34. 56d2861 Roll chromium_revision 33f8af5b42..8af2e3f3e3 (1629187:1629518) by chromium-webrtc-autoroll · 6 days ago
  35. 4df4ca4 Replace UsedPayloadTypes with PayloadTypeSuggester in CodecVendor. by Harald Alvestrand · 6 days ago
  36. d5848cf Add documentation for testing best practice in WebRTC by Harald Alvestrand · 6 days ago
  37. 1f95d4b Add Reported lost time series to ECN feedback graph in event log visualizer by Per K · 6 days ago
  38. c754e2f Roll chromium_revision 38a3b7e523..33f8af5b42 (1629041:1629187) by chromium-webrtc-autoroll · 6 days ago
  39. c1612b4 Configure RTCP mode during RTP/RTCP module construction by Tommi · 6 days ago
  40. babddf4 Reland "Detect codec collisions between audio and video sections" by Harald Alvestrand · 6 days ago
  41. fcaf816 Update WebRTC code version (2026-05-12T04:07:04). by webrtc-version-updater · 6 days ago
  42. 2342afb Roll chromium_revision cae7d1afb8..38a3b7e523 (1628845:1629041) by chromium-webrtc-autoroll · 6 days ago
  43. 6ee607f Roll chromium_revision 8d21585365..cae7d1afb8 (1628689:1628845) by chromium-webrtc-autoroll · 7 days ago
  44. 0aa676b Implement support machinery for payload type allocation redesign. by Harald Alvestrand · 7 days ago
  45. 119ae51 Roll chromium_revision 1f0c44e8c8..8d21585365 (1628456:1628689) by chromium-webrtc-autoroll · 7 days ago
  46. 38e88a1 Consolidate remote SSRC representation in audio receive components by Tommi · 7 days ago
  47. f406df7 Add IsEmpty to RtpStreamReceiverController and RtpDemuxer by Tommi · 7 days ago
  48. c751b57 Move integration test helper functions from .h to .cc by Harald Alvestrand · 7 days ago
  49. cd2de2c Roll chromium_revision eb721a86c0..1f0c44e8c8 (1628325:1628456) by chromium-webrtc-autoroll · 7 days ago
  50. 1b1ecff Reland "sdp: introduce MCD::AttributeLevel for session/media-level attrs" by Philipp Hancke · 7 days ago
  51. ed3c6e4 Disallow RTP header extension ID of 0 by Johannes Kron · 7 days ago
  52. b3a6073 Rename target_delay to stats_target_delay in VideoDelayTimings. by Åsa Persson · 7 days ago
  53. ca8bc89 Update WebRTC code version (2026-05-11T04:04:59). by webrtc-version-updater · 7 days ago
  54. 29073bb Roll chromium_revision 578404204d..eb721a86c0 (1628216:1628325) by chromium-webrtc-autoroll · 7 days ago
  55. ba7594a Update WebRTC code version (2026-05-10T04:07:40). by webrtc-version-updater · 8 days ago
  56. df25bc4 Roll chromium_revision 517c5eb895..578404204d (1628092:1628216) by chromium-webrtc-autoroll · 9 days ago
  57. e5ca0ed Revert "Detect codec collisions between audio and video sections" by Tomas Gunnarsson · 9 days ago
  58. f1424da Reland "Move signaling safety flag into SctpDataChannel and clarify its purpose" by Tommi · 9 days ago
  59. 078396f Delete workaround Thread implementation that do not set self as TaskQueue by Danil Chapovalov · 9 days ago
  60. 16ac3c3 Check validity of RTP header extenision ID at construction by Harald Alvestrand · 9 days ago
  61. b92fa6c Update WebRTC code version (2026-05-09T04:08:47). by webrtc-version-updater · 9 days ago
  62. 7ec277e Roll chromium_revision 42fb7fa15f..517c5eb895 (1627573:1628092) by chromium-webrtc-autoroll · 9 days ago
  63. a8aae89 Detect codec collisions between audio and video sections by Harald Alvestrand · 10 days ago
  64. b288713 Adds rust version of webrtc::RateTracker by Evan Shrubsole · 10 days ago
  65. 7e3a9ce Rely on TaskQueueBase interface in modules/rtp_rtcp by Danil Chapovalov · 10 days ago
  66. 5ab65fd Revert "sdp: introduce MCD::AttributeLevel for session/media-level attrs" by Johannes Kron · 10 days ago
  67. bb09586 Roll chromium_revision 330c8c76f5..42fb7fa15f (1627458:1627573) by chromium-webrtc-autoroll · 10 days ago
  68. bb7a22c Move MaxWaitingTime and associated state to FrameDecodeTiming. by Åsa Persson · 10 days ago
  69. 4331406 Update field-trials.md for clarity and freshness by Harald Alvestrand · 10 days ago
  70. 15383c5 Update Call::ReceiveStats to be associated with the network thread by Tommi · 10 days ago
  71. 60094aa sdp: introduce MCD::AttributeLevel for session/media-level attrs by Philipp Hancke · 10 days ago
  72. 93755bc Add a missing include on android by Nico Weber · 10 days ago
  73. 56fea02 Update WebRTC code version (2026-05-08T04:05:03). by webrtc-version-updater · 10 days ago
  74. 7d0cb6f Roll chromium_revision 74909b4add..330c8c76f5 (1627071:1627458) by chromium-webrtc-autoroll · 10 days ago
  75. 85a228f Roll chromium_revision a7864a7e22..74909b4add (1626575:1627071) by chromium-webrtc-autoroll · 11 days ago
  76. a4a2793 Revert "Move signaling safety flag into SctpDataChannel and clarify its purpose" by Johannes Kron · 11 days ago
  77. 0ea8047 Add OnFrameDropped override to vp9 encoder fuzzer. by Erik Språng · 11 days ago
  78. 81d438a Move signaling safety flag into SctpDataChannel and clarify its purpose by Tommi · 11 days ago
  79. 7372c48 Prevent wrong scalability mode from being used when base layer inactive. by Erik Språng · 11 days ago
  80. 1bec643 Use TimeController instead of FakeClock in fuzzers/RtpReplayer by Danil Chapovalov · 11 days ago
  81. 64f171a Refactor pc/media_session_unittest.cc and introduce Yoda-test swapping tool. by Harald Alvestrand · 11 days ago
  82. 5d46eb3 Remove rusty base64 implementation by Evan Shrubsole · 11 days ago
  83. 8638b13 LNA: Return after unexpected permission callback by Daniel.L (Byoungchan Lee) · 11 days ago
  84. 0751415 Reland "Add rust versions of Timestamp and TimeDelta" by Evan Shrubsole · 11 days ago
  85. 5e897066 Update WebRTC code version (2026-05-07T04:05:07). by webrtc-version-updater · 11 days ago
  86. 78e34f5 Roll chromium_revision 03b98044d1..a7864a7e22 (1626400:1626575) by chromium-webrtc-autoroll · 12 days ago
  87. d75b48f Roll chromium_revision 1c0643ce6a..03b98044d1 (1626148:1626400) by chromium-webrtc-autoroll · 12 days ago
  88. f9c143a ScreamV2, add application limited state by Per K · 12 days ago
  89. 40e217c Roll chromium_revision dc14bea0ed..1c0643ce6a (1625919:1626148) by chromium-webrtc-autoroll · 12 days ago
  90. c3e36a4 snap: add RTCConfiguration for enabling SNAP by Philipp Hancke · 12 days ago
  91. 4ed56ee Revert "Add rust versions of Timestamp and TimeDelta" by Evan Shrubsole · 12 days ago
  92. 9b4ffb3 In SctpDataChannel use plain bool as safety flag. by Danil Chapovalov · 12 days ago
  93. 46eda2f Remove arm32 bots and redundant arm64 bots by Jeremy Leconte · 12 days ago
  94. 748ae96 Remove redundant android_compile_arm64_rel bot from CQ by Jeremy Leconte · 12 days ago
  95. f4a58be Fix misformated tables in style guide by Danil Chapovalov · 12 days ago
  96. ab26421 Deprecate ArrayView alias by Danil Chapovalov · 12 days ago
  97. cc81bf9 Add rust versions of Timestamp and TimeDelta by Evan Shrubsole · 12 days ago
  98. f1e51c0 Activate corruption detection tests. by Emil Vardar · 12 days ago
  99. 42abd27 sdp munging: detect modification of msid stream/track by Philipp Hancke · 12 days ago
  100. 125a9d2 Wayland capture: validate buffer geometry before pixel copy by Jan Grulich · 12 days ago