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  1. 21c456e Update WebRTC code version (2024-09-10T04:06:53). by webrtc-version-updater · 4 hours ago main master
  2. 4ea6534 Roll chromium_revision c339b49443..4a8f19d868 (1353018:1353126) by chromium-webrtc-autoroll · 4 hours ago
  3. 110f7db Roll chromium_revision 33ef804c4e..c339b49443 (1352775:1353018) by chromium-webrtc-autoroll · 9 hours ago
  4. dc56a36 Use PayloadTypePicker in WebRtcVoiceEngine by Harald Alvestrand · 2 days ago
  5. 927244d Set MID in AudioReceiveChannel by Harald Alvestrand · 3 days ago
  6. 27db338 Roll chromium_revision c03ff62a28..33ef804c4e (1351560:1352775) by chromium-webrtc-autoroll · 16 hours ago
  7. 0f61f60 Mock call to os.path.isdir in roll_deps_test. by Björn Terelius · 18 hours ago
  8. 76aa330 Implement ObjCVideoEncoderFactory::QueryCodecSupport by Danil Chapovalov · 19 hours ago
  9. 0acbb77 Pass Environment into RtcpSender by Danil Chapovalov · 20 hours ago
  10. 363dc19 SimulcastToSvcConverter: Allow not setting scalability mode on frame by Ilya Nikolaevskiy · 20 hours ago
  11. 02113a2 Pass Environment into RtcpReceiver by Danil Chapovalov · 4 days ago
  12. 3652dd3 Review documentation and update review date by Artem Titov · 22 hours ago
  13. 65b59a9 Prepend webrtc ns to StrJoin calls in dcsctp ns by Dor Hen · 2 days ago
  14. 26146bb Add support for screencast with temporal layering to SvcRateAllocator by Sergey Silkin · 23 hours ago
  15. 405f343 Update WebRTC code version (2024-09-07T04:08:12). by webrtc-version-updater · 3 days ago
  16. 6f64ae1 Extract corruption detection message to its own target by Fanny Linderborg · 4 days ago
  17. 65b46da dcsctp: Don't send FORWARD-TSN in its own chunk by Victor Boivie · 4 days ago
  18. 7929ef5 dcsctp: Add test for stream reset pending by Victor Boivie · 8 days ago
  19. c9aaf11 Remove use of AcmReceiver in ChannelReceive by Henrik Lundin · 4 days ago
  20. 3ad2c8d Make getNumObservers @VisibleForTesting so that it can be tested outside of package org.webrtc by Jonas Oreland · 4 days ago
  21. 9f096a8 Allow VideoEncoderSoftwareFallbackWrapper to return SIMULCAST_PARAMS_NOT_SUPPORTED by Ilya Nikolaevskiy · 4 days ago
  22. c7da857 Fix lint issues in pacing/ by Björn Terelius · 6 days ago
  23. f92f39e Increase the default maximum jitter buffer size to 200 packets for Android. by karllen.zheng@ringcentral.com · 4 days ago
  24. 4334cdf Reland "Return audio stats regarless if we have a codec." by Jakob Ivarsson · 5 days ago
  25. 5913803 Update WebRTC code version (2024-09-06T04:04:53). by webrtc-version-updater · 4 days ago
  26. f5c5fb9 Roll chromium_revision 040c638bdb..c03ff62a28 (1351313:1351560) by chromium-webrtc-autoroll · 5 days ago
  27. e922cd1 Use Environment instead of Clock in ModuleRtpRtcp and its RTP subcomponents by Danil Chapovalov · 5 days ago lkgr
  28. e94c7da Revert "Return audio stats regarless if we have a codec." by Jakob Ivarsson‎ · 5 days ago
  29. 7fff587 Return audio stats regarless if we have a codec. by Jakob Ivarsson · 5 days ago
  30. 5162dc3 Reland "TaskQueueStdlib: Stop spamming on idle." by Markus Handell · 5 days ago
  31. 5ac7495 Prepare to use SimulcastToSvcConverter in chromium by Ilya Nikolaevskiy · 5 days ago
  32. d4c5843 Undo recent changes to initial frame dropper, fixing a regression. by Henrik Boström · 5 days ago
  33. 9463095 Revert "TaskQueueStdlib: Stop spamming on idle." by Markus Handell · 5 days ago
  34. 6255a7f Avoid negative timestamp in SourceTracker. by Jakob Ivarsson · 6 days ago
  35. 3047e64 Roll chromium_revision 7b9940f0b2..040c638bdb (1351213:1351313) by chromium-webrtc-autoroll · 5 days ago
  36. 05e1a1f Explicitly set encoder_context to nullptr in SimulcastEncoderAdapter after move. by Björn Terelius · 5 days ago
  37. 010c189 Move thread handling from source tracker. by Jakob Ivarsson · 6 days ago
  38. 2fbaa8e TaskQueueStdlib: Stop spamming on idle. by Markus Handell · 5 days ago
  39. 3144e20 Make build_helpers.py work in the chromium/src superproject by Gavin Mak · 5 days ago
  40. b2a2b1b Update WebRTC code version (2024-09-05T04:05:51). by webrtc-version-updater · 5 days ago
  41. e89abdc Roll chromium_revision 524f525d0b..7b9940f0b2 (1351076:1351213) by chromium-webrtc-autoroll · 5 days ago
  42. 55ca3c4 Roll chromium_revision 12d29901e2..524f525d0b (1350835:1351076) by chromium-webrtc-autoroll · 5 days ago
  43. 8480808 ssl: remove SSL_set_read_ahead for DTLS mode by Philipp Hancke · 9 days ago
  44. af8f626 Use Environment instead of Clock in ModuleRtpRtcp2 and its RTP subcomponents by Danil Chapovalov · 6 days ago
  45. d36041e Roll chromium_revision 621a1d6f77..12d29901e2 (1350725:1350835) by chromium-webrtc-autoroll · 6 days ago
  46. ac505c5 Enable the FrameInstrumentationGenerator if its extension is negotiated by Fanny Linderborg · 6 days ago
  47. 6e5eaea Roll chromium_revision 66e155a442..621a1d6f77 (1350363:1350725) by Björn Terelius · 6 days ago
  48. 70a59b6 Flip default value of AndroidNetworkMonitor field trials by Jonas Oreland · 6 days ago
  49. e540648 Remove trailing semicolons in Java by Björn Terelius · 7 days ago
  50. 2da07c8 Update docs about supported platforms and compilers. by Mirko Bonadei · 6 days ago
  51. 64d68c3 Add WebRTC-MixedCodecSimulcast field trial by Florent Castelli · 7 days ago
  52. 4a7ea89 Fix lint issues in logging/ by Björn Terelius · 7 days ago
  53. dac0805 Add FrameInstrumentationData to RTPVideoHeader and CodecSpecificInfo by Fanny Linderborg · 7 days ago
  54. 55a5933 Minor format to extrapolate local time by yazdan0a · 7 days ago
  55. 45065a7 Delete deprecated AudioDecoderFactory::MakeAudioDecoder by Danil Chapovalov · 7 days ago
  56. 40a038e Update WebRTC code version (2024-09-04T04:08:21). by webrtc-version-updater · 6 days ago
  57. ada1720 Roll chromium_revision 01d6daf051..66e155a442 (1350197:1350363) by chromium-webrtc-autoroll · 6 days ago
  58. 0c2cd62 Fix lint issues in congestion_controller. by Björn Terelius · 7 days ago
  59. f8cb8b7 Roll chromium_revision cae6b92cf5..01d6daf051 (1349874:1350197) by chromium-webrtc-autoroll · 7 days ago
  60. c17ca01 Move the payload type picker to call/ by Harald Alvestrand · 7 days ago
  61. 682f794 Deprecate bad signature for CreateSessionDescription. by Kári Tristan Helgason · 11 days ago
  62. e432503 Rewrite simulcast config to equivalent SVC for vp9 simulcast by Ilya Nikolaevskiy · 7 days ago
  63. fb7c306 Run include cleaner on subset of modules/rtp_rtcp by Danil Chapovalov · 8 days ago
  64. c5b9a60 Propagate environment to RtpSenders by Florent Castelli · 7 days ago
  65. 8401f56 Add fieldtrials WebRTC-QCM-Static-{AV1, VP8, VP9} by Johannes Kron · 7 days ago
  66. 3d60f25 Fix gtest/gmock includes in apply-include-cleaner script. by Jeremy Leconte · 7 days ago
  67. 3881cb6 PipeWire camera: make member variable with the PipeWire status updated by Jan Grulich · 8 days ago
  68. 863c2c9 Roll chromium_revision b975bdde27..cae6b92cf5 (1348475:1349874) by Björn Terelius · 8 days ago
  69. 6e072e6 Rename is_key_frame to communicate_upper_bits in FrameInstrumentation*Data by Fanny Linderborg · 7 days ago
  70. 843a317 Fix requested_resolution orientation assumption in OnSinkWants(). by Henrik Boström · 7 days ago
  71. d34f3b8 Remove more self assignment in if-clause by Bjorn Terelius · 7 days ago
  72. 93c9aa1 Apply include-cleaner to call/ by Harald Alvestrand · 7 days ago
  73. 5eb8588 Move FrameInstrumentation*Data structs to common_video by Fanny Linderborg · 8 days ago
  74. a82eb4e Remove self assignment in if-clause by Björn Terelius · 8 days ago
  75. 55ed950 Propagate corruption score to VideoReceiverInfo. by Emil Vardar · 8 days ago
  76. 99874e7 Update WebRTC code version (2024-09-03T04:04:21). by webrtc-version-updater · 7 days ago
  77. 77eba46 Adding ChannelStatistics Logs by Daniel · 7 days ago
  78. 86251a0 rewrite SSLInfoCallback logging by Philipp Hancke · 2 weeks ago
  79. 04ab497 Review abseil-in-webrtc for freshness by Danil Chapovalov · 8 days ago
  80. 86ac1df Fix libsrtp openssl build by Philipp Hancke · 14 days ago
  81. 9212f09 Update Abseil instructions for absl::optional by Florent Castelli · 12 days ago
  82. 8037fc6 Migrate absl::optional to std::optional by Florent Castelli · 12 days ago
  83. 787b907 Update freshness of the h-cc-pairs section of the style guide by Danil Chapovalov · 8 days ago
  84. 4e41db2 Propagate Environment to RtpRtcp module in FlexfecReceiver by Danil Chapovalov · 11 days ago
  85. 164b3b3 Introduce ModuleRtpRtcpImpl factory that accepts Environment by Danil Chapovalov · 11 days ago
  86. cb00e16 Revert "Enable 'iwyu_verifier' bot." by Jeremy Leconte · 8 days ago
  87. af7155e Propagate Environment to video RtpRtcp modules by Danil Chapovalov · 11 days ago
  88. 5a92ddb Updates review date in ADM g3doc. by henrika · 8 days ago
  89. 24366b0 Propagate Environment to audio RtpRtcp modules by Danil Chapovalov · 11 days ago
  90. 0b4b5b0 Use AV1E_SET_AUTO_TILES by Sergey Silkin · 11 days ago
  91. a4cf34d Enable 'iwyu_verifier' bot. by Jeremy Leconte · 8 days ago
  92. dd86c95 Update WebRTC code version (2024-09-02T04:06:36). by webrtc-version-updater · 8 days ago
  93. 177788f Update WebRTC code version (2024-09-01T04:05:33). by webrtc-version-updater · 9 days ago
  94. 91eacf3 Update WebRTC code version (2024-08-31T04:05:52). by webrtc-version-updater · 10 days ago
  95. 738abe0 Upgrade ios version used for perf tests. by Jeremy Leconte · 11 days ago
  96. c4d7493 Add some flags to 'apply-include-cleaner'. by Jeremy Leconte · 11 days ago
  97. d385af5 Introduce ModuleRtpRtcpImpl2 constructor that accepts Environment by Danil Chapovalov · 12 days ago
  98. 058972f Make LAYER_DROP and max_consec_drop=2 to be default settings by Sergey Silkin · 12 days ago
  99. b5f4006 Inject field trials in NetEqTest instead of setting global. by Jakob Ivarsson · 4 weeks ago
  100. 8d478dd Roll chromium_revision 10ff7fa1e3..b975bdde27 by Jeremy Leconte · 11 days ago