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  1. f382b0d Update WebRTC code version (2026-02-15T04:07:17). by webrtc-version-updater · 18 hours ago main master
  2. 25713d9 Separate transceiver references from the stats info struct by Tommi · 36 hours ago lkgr
  3. 5e1c815 Remove deprecated --no_auth flag from DEPS by Bjorn Terelius · 2 days ago
  4. e952c94 Update WebRTC code version (2026-02-14T04:06:46). by webrtc-version-updater · 2 days ago
  5. cc70943 Migrate internal usages of ByteOrder to ArrayView API by Harald Alvestrand · 2 days ago
  6. a3c1ddd Migrate from ArrayView::subview to subspan in audio code by Danil Chapovalov · 2 days ago
  7. 60c1862 Fork Chromium's TestStatusReporter to avoid a dependency on //base. by Mirko Bonadei · 2 days ago
  8. f81c7f5 Spanify some files in pc/ by Harald Alvestrand · 2 days ago
  9. e27b7b6 Remove reference to catapult's github repository by Victor Hugo Vianna Silva · 2 days ago
  10. 6d4ca81 Reland "Split analyzer.cc and change CreateFooGraph methods to free functions" by Bjorn Terelius · 2 days ago
  11. 8ca627d Fix potential deadlock in RTCStatsCollector during shutdown. by Tommi · 2 days ago
  12. 15fd636 [tracing] Migrate legacy async macros in webrtc by Etienne Pierre-doray · 2 days ago
  13. cf7d2b1 Add support for OnFrameDropped callback in FrameEncodeMetadataWriter. by Erik Språng · 3 days ago
  14. a4e0209 Video timing simulator: Implement stream metrics. by Rasmus Brandt · 3 days ago
  15. 48d38b6 Removes WebRTC.DesktopCapture.Win.WgcDirtyRegionSupport UMA by henrika · 3 days ago
  16. a6c2896 Revert "Split analyzer.cc and change CreateFooGraph methods to free functions" by Björn Terelius · 3 days ago
  17. 68cb390 Split analyzer.cc and change CreateFooGraph methods to free functions by Bjorn Terelius · 3 days ago
  18. a7e7130 Deprecate the two-argument Buffer constructor by Harald Alvestrand · 3 days ago
  19. 191a6ab Batch worker thread tasks in [Get]StopTransceiverProcedure by Tommi · 3 days ago
  20. e1357d9 Declare rtc_base/buffer safe from unsafe_buffers_paths by Harald Alvestrand · 3 days ago
  21. 6fe2890 Add payload_type.h to API by Harald Alvestrand · 3 days ago
  22. 4cff6ae Revert "Fix std::tmpnam deprecation in event_log_visualizer" by Tomas Gunnarsson · 3 days ago
  23. 1ccc59d Spanify CopyOnWriteBuffer by Danil Chapovalov · 4 days ago
  24. 814e92f Switch to python3 as default script_executable. by Mirko Bonadei · 4 days ago
  25. 20101b2 Fix std::tmpnam deprecation in event_log_visualizer by AhmadDurrani579 · 4 days ago
  26. 347fb3e docs: Add pointer nullability annotation guidelines by Tommi · 4 days ago
  27. a62566a Update WebRTC code version (2026-02-12T04:07:43). by webrtc-version-updater · 4 days ago
  28. ccd1d88 Roll chromium_revision 687500049d..cabe765c34 (1583344:1583505) by chromium-webrtc-autoroll · 4 days ago
  29. 5700d8f Re-enable ios debug simulator bot. by Jeremy Leconte · 4 days ago
  30. a5a020f Fix VideoStreamEncoderTest.QualityScalerTriggersOnFrameDropping on iOS by Erik Språng · 4 days ago
  31. 0ab329d Roll chromium_revision 27f57c3b13..687500049d (1583238:1583344) by chromium-webrtc-autoroll · 4 days ago
  32. be9a885 Roll chromium_revision 974282c604..27f57c3b13 (1583127:1583238) by chromium-webrtc-autoroll · 4 days ago
  33. cc688f5 Use Clock and RunLoop in null_socket_server_unittest by Evan Shrubsole · 4 days ago
  34. 3d02773 Change logging severity to LS_VERBOSE for `RenderingTracker::OnDecodableFrameTimeout` by Rasmus Brandt · 4 days ago
  35. d0aec8f Video timing simulator: Add SSRC filtering to `RtcEventLogDriver` by Rasmus Brandt · 4 days ago
  36. 27eddf0 Use clock in Logging Perf test by Evan Shrubsole · 4 days ago
  37. 5738f01 Roll chromium_revision be8f64a423..974282c604 (1582802:1583127) by chromium-webrtc-autoroll · 4 days ago
  38. 7ce2210 Use clock in rtc_event_log_unittest_helper by Evan Shrubsole · 5 days ago
  39. 8485ffd Add network queue to the schedulable network. by Jeremy Leconte · 5 days ago
  40. cb94a77 Video timing simulator: Improve per-frame margin metrics. by Rasmus Brandt · 5 days ago
  41. 964afe5 Replace ArrayView::subview with subspan in H26x parsers by Danil Chapovalov · 5 days ago
  42. a5e2e7b video_capture: move capture_checker check after StopCapture in V4L2 dtor by Nfrederiksen · 5 days ago
  43. 90c4b85 Batch SetupMediaChannel blocking calls in rtp receiver classes. by Tommi · 5 days ago
  44. 43845d3 dcsctp: Correct Fast Recovery retransmission logic by Victor Boivie · 5 days ago
  45. b9336fd Update WebRTC code version (2026-02-11T04:02:53). by webrtc-version-updater · 5 days ago
  46. b3bdddb Roll chromium_revision dbeec2f05b..be8f64a423 (1580859:1582802) by chromium-webrtc-autoroll · 5 days ago
  47. e902e15 Add network slice to Network::ToString() by Sameer Vijaykar · 5 days ago
  48. 7fa36c5 Transition quality scaler to new DropReason type. by Erik Språng · 5 days ago
  49. dc0041e Add RTC_CHECK to fwrite calls in video_file_writer by AhmadDurrani579 · 5 days ago
  50. a27d608 Spanify rtc_base/buffer by Harald Alvestrand · 5 days ago
  51. d01ecda Replace ArrayView::subview with subspan in logging/ by Danil Chapovalov · 5 days ago
  52. 7c388cb Changing default values of multi channel processing to be true by Per Åhgren · 6 days ago
  53. ffe7987 Update WebRTC code version (2026-02-10T04:08:07). by webrtc-version-updater · 6 days ago
  54. 573e746 Correct the switching between the coarse and refined filters by Per Åhgren · 6 days ago
  55. d1972ad Fix potential division by zero in OpenYuvFile by AhmadDurrani579 · 6 days ago
  56. 24e5ff8 Add UMA histograms for PSNR stats. by Erik Språng · 6 days ago
  57. 6fa537a Use callbacks and RunLoop in PeerConnection ICE tests by Tommi · 6 days ago
  58. 229bda0 Flip `RenderingSimulator::reuse_streams` to default false. by Rasmus Brandt · 6 days ago
  59. 297352a Update comfort noise to be applied identically on all channels by Per Åhgren · 6 days ago
  60. a0d5da4 Refactor PeerConnectionIntegrationWrapper::NewGetStats to use RunLoop by Tommi · 6 days ago
  61. 54e6613 SEA: Add support for OnFrameDropped and end_of_temporal_unit flag. by Erik Språng · 6 days ago
  62. 28987e4 Fix undefined behavior in Y4M header parsing by AhmadDurrani579 · 6 days ago
  63. 5aa2cd6 Revert "Use injected clock in android_video_track_source" by Evan Shrubsole · 7 days ago
  64. a6d83994 Remove aec_mobile (AECm) config from audio_processing.h by Lionel Koenig Gélas · 7 days ago
  65. 3d19cb9 Use clock from Environment in direct_transport by Evan Shrubsole · 7 days ago
  66. 3b4a327 Use injected clock in android_video_track_source by Evan Shrubsole · 7 days ago
  67. 1d1ee45 Refactor AudioProcessingTest to match current audio processing code by Per Åhgren · 7 days ago
  68. 066cbff Update WebRTC code version (2026-02-09T04:06:40). by webrtc-version-updater · 7 days ago
  69. 4520c78 Pass Clock to BitrateAdjuster at construction by Danil Chapovalov · 7 days ago
  70. 3a9fe16 Update WebRTC code version (2026-02-08T04:04:50). by webrtc-version-updater · 8 days ago
  71. 62f5e23 Update WebRTC code version (2026-02-07T04:04:55). by webrtc-version-updater · 9 days ago
  72. afbb1d2 Add histograms for AV1 and H265 QPs. by Erik Språng · 9 days ago
  73. a13738f Update root OWNERS by Fredrik Solenberg · 9 days ago
  74. 6e2cbdf Roll chromium_revision 2d06e1566f..dbeec2f05b (1580686:1580859) by chromium-webrtc-autoroll · 9 days ago
  75. 82d4334 rtc_tools: Fix build error in chromium builders by Takuto Ikuta · 9 days ago
  76. eee78e7 Reland "Use Clock in internal EventTracer" by Evan Shrubsole · 9 days ago
  77. 9a7fadf Replace ArrayView::subview with subspan in modules/rtp_rtcp by Danil Chapovalov · 9 days ago
  78. df428a3 Video timing simulator: Add `ResultsBase` with `IsEmpty()` helper. by Rasmus Brandt · 10 days ago
  79. d6275c2 Roll chromium_revision 9201f0ec67..2d06e1566f (1580225:1580686) by chromium-webrtc-autoroll · 10 days ago
  80. cee4e29 Replace ArrayView::subview usage with subspan in dcsctp by Danil Chapovalov · 10 days ago
  81. 94ce396 Update whitespace by Evan Shrubsole · 10 days ago
  82. 770eedb Default enable fixed delay mode NACK for audio. by Jakob Ivarsson · 10 days ago
  83. 3f4d30b Revert "Use Clock in internal EventTracer" by Tomas Gunnarsson · 10 days ago
  84. 86e1eb8 Update WebRTC code version (2026-02-06T04:04:51). by webrtc-version-updater · 10 days ago
  85. 6952fae Reduce use of deprecated GetStats implementation. by Tommi · 10 days ago
  86. 9c0ab1e Roll chromium_revision 4da100fca9..9201f0ec67 (1580121:1580225) by chromium-webrtc-autoroll · 10 days ago
  87. b28858d Use SimulatedClock in timestamp_aligner_unittest by Evan Shrubsole · 10 days ago
  88. 1d48dd3 ML residual echo estimation: explicitly set TF Lite num threads by Sam Zackrisson · 10 days ago
  89. ed6507b Use Clock in internal EventTracer by Evan Shrubsole · 10 days ago
  90. a2b172a Roll chromium_revision 1fb3d7f881..4da100fca9 (1580004:1580121) by chromium-webrtc-autoroll · 10 days ago
  91. 250f248 Mark old version of GetStats() as deprecated by Tommi · 10 days ago
  92. 33387a2 Move more RTC stats production to signaling thread. by Tommi · 10 days ago
  93. a4ea3c2 Roll chromium_revision fcfac780b8..1fb3d7f881 (1568120:1580004) by chromium-webrtc-autoroll · 11 days ago
  94. e1373fb dcsctp: Correct outstanding data calculation by Victor Boivie · 11 days ago
  95. 6962bb0 Ignore testBluetoothScoAudioStartAndStop by Evan Shrubsole · 11 days ago
  96. ce5c33f Delete TransportFeedbackObserver interface as unused by Danil Chapovalov · 11 days ago
  97. 677c408 Update whitespace by Tommi · 11 days ago
  98. c2cc651 Update WebRTC code version (2026-02-05T04:04:01). by webrtc-version-updater · 11 days ago
  99. c6283a2 Revert "Reland "Activate adaptive channel mixing when the echo canceller runs in stereo"" by Tomas Gunnarsson · 11 days ago
  100. e7702dc Inline unique_ptr field(s) to improve data locality. by Jeremy Leconte · 11 days ago