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webrtc/src/HEAD
  1. 126f7fa build: clean up rtc_base deps by Philipp Hancke · 7 minutes ago main master
  2. f201b69 Batch resetting of unsignaled receive streams by Tommi · 5 hours ago lkgr
  3. d7181bf dcsctp: Resend SHUTDOWN ACK on timer expiration by Victor Boivie · 11 hours ago
  4. 4cfcc2d Stabilize RenegotiateManyVideoTransceiversAndWatchAudioDelay test by Tommi · 11 hours ago
  5. 774a7fc [ObjC SDK]: Added scalabilityMode to RTCRtpEncodingParameters by Jury Jaraševič · 16 hours ago
  6. 8194628 Replace task vectors with ScopedOperationsBatcher by Tommi · 17 hours ago
  7. f5decab Update thread blocking call metrics and batcher logging by Tommi · 18 hours ago
  8. 6df60e8 Revert "Deprecate ArrayView alias" by Philip Eliasson · 20 hours ago
  9. 216c45c Merge logic from RenderTimeInternal into public RenderTime. by Åsa Persson · 20 hours ago
  10. 2baeac4 Move NetworkSlice enum to network_constants. by Sameer Vijaykar · 20 hours ago
  11. ddb37ff Revert "Fix inlining of ArrayView<T>" by Danil Chapovalov · 21 hours ago
  12. 23cba81 Fix inlining of ArrayView<T> by Danil Chapovalov · 23 hours ago
  13. 6c2e98d Update WebRTC code version (2026-04-07T04:08:38). by webrtc-version-updater · 27 hours ago
  14. e3db085 Default enable DTLS1.3 by Jonas Oreland · 2 days ago
  15. d6fe9f1 Update WebRTC code version (2026-04-06T04:09:15). by webrtc-version-updater · 2 days ago
  16. 705ee08 Refresh h-cc-pairs style doc by Danil Chapovalov · 3 days ago
  17. 991cfe3 Update WebRTC code version (2026-04-04T04:08:02). by webrtc-version-updater · 4 days ago
  18. 15c9492 Update WebRTC code version (2026-04-03T04:05:14). by webrtc-version-updater · 5 days ago
  19. 16bb4fc Reword 'how to depend on Abseil' section by Danil Chapovalov · 6 days ago
  20. af15fac Deprecate ArrayView alias by Danil Chapovalov · 6 days ago
  21. ca896b7 Fix rebase issue of two colliding CLs by Tommi · 6 days ago
  22. a2bd172 Move payload type demuxing management to RtpTransport by Tommi · 6 days ago
  23. 2432b69 Capture channel pointers on signaling thread to fix data race by Tommi · 6 days ago
  24. 2b7dced Use RunLoop in audio/ tests by Evan Shrubsole · 6 days ago
  25. 52e8522 Replace ArrayView with std::span in api/ by Danil Chapovalov · 6 days ago
  26. c6b486d Update WebRTC code version (2026-04-02T04:08:10). by webrtc-version-updater · 6 days ago
  27. f2b4180 Allow EventLogAnalyzer to be created without a valid parsed log by Per K · 7 days ago
  28. 2dab14a Use RunLoop in call/ tests by Evan Shrubsole · 7 days ago
  29. 06d113a Use RunLoop in p2p/ tests by Evan Shrubsole · 7 days ago
  30. 4f39938 Use real Thread in examples/ instead of AutoThread by Evan Shrubsole · 7 days ago
  31. 6b3d4ef Use RunLoop in media/ tests by Evan Shrubsole · 7 days ago
  32. 507cc3d Use RunLoop in video/ tests by Evan Shrubsole · 7 days ago
  33. 6815c67 Use RunLoop in rtc_base/ tests by Evan Shrubsole · 7 days ago
  34. 6331862 Group members in VCMTiming into a VideoDelayTimings struct. by Åsa Persson · 7 days ago
  35. 9f086f0 Use RunLoop in test/ tests by Evan Shrubsole · 7 days ago
  36. ae72a71 Use RunLoop in pc/ tests by Evan Shrubsole · 7 days ago
  37. e10cd19 Audio: Enable dynamic on-the-fly AEC3 configuration updates for the suppressor gain computation by Jesús de Vicente Peña · 7 days ago
  38. 5a90c2d Delete deprecated reinterpret_array_view by Danil Chapovalov · 7 days ago
  39. 0ead3cb Use ScopedOperationsBatcher for media description pushdown by Tommi · 7 days ago
  40. 6075b7e Add gn-check-autofix skill by Evan Shrubsole · 8 days ago
  41. 3f11f14 Update WebRTC code version (2026-03-31T04:10:44). by webrtc-version-updater · 8 days ago
  42. f79b115 Replace ArrayView with std::span in modules/video_coding/av1 by Danil Chapovalov · 8 days ago
  43. f1c5550 Move the LibaomAv1Encoder V2 encoder implementation to separate file. by Erik Språng · 9 days ago
  44. 36def04 Scream, ensure feedback contains at least one received packet by Per K · 9 days ago
  45. 443e7ad Synchronize final stats collection in E2E quality tests by Tommi · 9 days ago
  46. c975291 Replace ArrayView with std::span everywhere except api by Danil Chapovalov · 9 days ago
  47. 50f35f0 Reapply "build: clean up modules audio build files" by Philipp Hancke · 9 days ago
  48. 65dd518 Update WebRTC code version (2026-03-30T04:12:15). by webrtc-version-updater · 9 days ago
  49. a73939f Update WebRTC code version (2026-03-29T04:05:05). by webrtc-version-updater · 10 days ago
  50. 4487631 PipeWire capture: fix camera capability enumeration race and duplicates by Jan Grulich · 11 days ago
  51. 876820e Update WebRTC code version (2026-03-28T04:04:44). by webrtc-version-updater · 11 days ago
  52. c25f3da Update ScopedOperationsBatcher tasks with finalizers to propagate errors by Tommi · 12 days ago
  53. 978f78b Support custom socket server in test::RunLoop by Evan Shrubsole · 12 days ago
  54. 830e031 Use the PayloadType class rather than uint8_t in Codec class by Harald Alvestrand · 12 days ago
  55. 39fcc21 Replace ArrayView with std::span in modules/ by Danil Chapovalov · 12 days ago
  56. fe2da40 Audio: Reduce max channels to 16 to prevent buffer overflow by Tommi · 12 days ago
  57. fd2e7ea Fix potential dcSCTP NackBetweenAckBlocks OOB access by Tommi · 12 days ago
  58. 190a780 p2ptc: clean up unused legacy API by Philipp Hancke · 12 days ago
  59. 5398785 Convert kMax constants in video_codec_constants to size_t by Harald Alvestrand · 12 days ago
  60. 2395c1f Remove SIMULATED_WAIT by Evan Shrubsole · 12 days ago
  61. c6ef49b More spring cleaning by Tommi · 12 days ago
  62. 0085c1c Rename methods in ScopedOperationsBatcher by Tommi · 12 days ago
  63. b83ca3e Create include-cleaner skill by Evan Shrubsole · 12 days ago
  64. c95a4a9 Add .gemini to .gitignore by Evan Shrubsole · 12 days ago
  65. fef1bf4 Refactor RTP Header Extension Management into RtpTransport by Tommi · 12 days ago
  66. 47601ff Replace ArrayView with std::span in common_video/ by Danil Chapovalov · 12 days ago
  67. b466437 Use first_ssrc() instead of ssrcs[0] in channel.cc by Tommi · 12 days ago
  68. b188c9f Update WebRTC code version (2026-03-27T04:10:01). by webrtc-version-updater · 12 days ago
  69. 4070f12 Use GlobalSimulatedTimeController in PortTest by Evan Shrubsole · 12 days ago
  70. 1f9fdd2 Replace ArrayView with std::span in modules/audio_processing by Danil Chapovalov · 13 days ago
  71. bb96cba Spring cleanup of WATCHLIST and OWNER files by Tommi · 13 days ago
  72. 0594661 Revert "build: clean up modules audio build files" by Mirko Bonadei · 13 days ago
  73. 989cdab Delete the AEC3 json config fuzzer by Sam Zackrisson · 13 days ago
  74. 5560f13 Truncate CSRC list in RtpPacket::SetCsrcs if too large by Tommi · 13 days ago
  75. 18f6a99 Change how TimingFrameInfo is passed to ReceiveStatisticsProxy. by Åsa Persson · 13 days ago
  76. 1b9f86f build: remove unnecessary targets in video_coding by Philipp Hancke · 13 days ago
  77. 246b8de build: clean up modules audio build files by Philipp Hancke · 13 days ago
  78. a32d5d8 Roll chromium_revision 5b78ada5e8..a3f5fcb392 (1604779:1604961) by chromium-webrtc-autoroll · 14 days ago
  79. 80dd204 Roll chromium_revision d3079a981f..5b78ada5e8 (1604639:1604779) by chromium-webrtc-autoroll · 14 days ago
  80. 84c9fda Initialize NetEqNetworkStatistics fields. by Jakob Ivarsson · 14 days ago
  81. 8551776 Add gtest-parallel skill by Evan Shrubsole · 14 days ago
  82. da927d7 Refactor stun_port_unittest to use GlobalSimulatedTimeController by Evan Shrubsole · 14 days ago
  83. 228f04d pc: rename data_channel_unittest by Philipp Hancke · 14 days ago
  84. 3d3d455 Correct setting Scream padding end time by Per K · 14 days ago
  85. 746e3f9 Use GlobalSimulatedTimeController in P2PTransportChannelTest by Evan Shrubsole · 14 days ago
  86. 1d5a6e1 Use GPU texture in desktop capture by Zhibo Wang · 14 days ago
  87. 9f4a516 Use subset profile matching for H.264 decoder support by Shaofan Qi · 2 weeks ago
  88. 277d8b2 Roll chromium_revision 60f1c8a981..d3079a981f (1604486:1604639) by chromium-webrtc-autoroll · 2 weeks ago
  89. 5e177d2 Harden SimpleStringBuilder safety checks by Tommi · 2 weeks ago
  90. a875e8c build: clean up api/BUILD.gn libjingle_peerconnection_api target by Jeremy Leconte · 2 weeks ago
  91. 9c36e11 Simplify thread yielding by removing support for high-priority tasks by Tommi · 2 weeks ago
  92. a08f6ce Update WebRTC code version (2026-03-25T04:04:33). by webrtc-version-updater · 2 weeks ago
  93. f85d783 Roll chromium_revision 8327d89a47..60f1c8a981 (1604334:1604486) by chromium-webrtc-autoroll · 2 weeks ago
  94. 13e7eba Roll chromium_revision 4acdf7133f..8327d89a47 (1604168:1604334) by chromium-webrtc-autoroll · 2 weeks ago
  95. 0bc96c8 Reland "Allow unspecified max allocatable bitrate in VideoSendStreamImpl" by Erik Språng · 2 weeks ago
  96. f4bfe0b Roll chromium_revision ba4fdb3ab0..4acdf7133f (1604035:1604168) by chromium-webrtc-autoroll · 2 weeks ago
  97. c397ee9 Follow-up auto -> explicit type by Tommi · 2 weeks ago
  98. 731795b Refactor AddCertificateReports to prevent crash by Tommi · 2 weeks ago
  99. 49a0591 Roll chromium_revision c9a20fd775..ba4fdb3ab0 (1603923:1604035) by chromium-webrtc-autoroll · 2 weeks ago
  100. e68d24d Replace LOG_AND_RETURN_ERROR with "return LOG_ERROR" by Harald Alvestrand · 2 weeks ago