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  1. cfd4cd0 Introduce AddDefaultRecvStreamForTesting to VideoReceiveChannel API by Harald Alvestrand · 11 months ago
  2. 5c35d08 Replace "RTRR" with "RRTR" by Philipp Hancke · 11 months ago
  3. 1ecac13 Roll chromium_revision be3e47cd99..aae661725b (1148555:1148994) by chromium-webrtc-autoroll · 11 months ago
  4. f4d0a49 Adopt EglThread in EglRenderer by Linus Nilsson · 11 months ago
  5. 1cc41ea Remove unused Win32Window class by Philipp Hancke · 11 months ago
  6. 0f13765 Delete RTC[NonStandard/Restricted]StatsMember. by Henrik Boström · 11 months ago
  7. 621cb29 Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. by Rasmus Brandt · 11 months ago
  8. 8ac66a2 Update WebRTC code version (2023-05-25T04:12:48). by webrtc-version-updater · 11 months ago
  9. ec863a2 Roll chromium_revision 8f46ad499d..be3e47cd99 (1148441:1148555) by chromium-webrtc-autoroll · 11 months ago
  10. 979b047 Revert "Temporarily add dummy trackId to unblock roll." by Henrik Boström · 11 months ago
  11. aa1ad7d In RtcpTransciever refactor outgoing transport interface by Danil Chapovalov · 11 months ago
  12. f4d762e docs: explain release note process by Philipp Hancke · 11 months ago
  13. 2057d71 [Stats] Delete unused NonStandardGroupId. by Henrik Boström · 11 months ago
  14. 61bacd1 Enable WebRTC-SplitMediaChannel by default by Harald Alvestrand · 12 months ago
  15. 3df4178 Temporarily add dummy trackId to unblock roll. by Henrik Boström · 11 months ago
  16. 4e231ee Delete deprecated 'track' and 'stream' metrics from WebRTC. by Henrik Boström · 11 months ago
  17. 54c37a5 Roll chromium_revision 9378e3160b..8f46ad499d (1148314:1148441) by chromium-webrtc-autoroll · 11 months ago
  18. 18898d7 Update WebRTC code version (2023-05-24T04:17:39). by webrtc-version-updater · 11 months ago
  19. 1920430 Roll chromium_revision 3d8a0c6a22..9378e3160b (1148203:1148314) by chromium-webrtc-autoroll · 11 months ago
  20. 531383c Roll chromium_revision bbf4ff3290..3d8a0c6a22 (1148035:1148203) by chromium-webrtc-autoroll · 11 months ago
  21. 4daa4e6 Roll chromium_revision c025c4ac7b..bbf4ff3290 (1147854:1148035) by chromium-webrtc-autoroll · 11 months ago
  22. a7d1081 Revert "pipewire capturer: Reduce the amount of copying" by Alexander Cooper · 11 months ago
  23. 2eacbbc Roll chromium_revision 692840e030..c025c4ac7b (1147747:1147854) by chromium-webrtc-autoroll · 11 months ago
  24. 33697291 Add EglThread class wrapping EglConnection and handler. by Linus Nilsson · 11 months ago
  25. ff35a37 Unit tests for MediaChannel creation API by Harald Alvestrand · 11 months ago
  26. 98f47a3 Delete redundant member StreamDataCounters::last_packet_received_time by Danil Chapovalov · 11 months ago
  27. 0328190 Add video_codec_perf_tests to desktop and android perf test suites by Sergey Silkin · 11 months ago
  28. e3441ec Roll chromium_revision fb6508249a..692840e030 (1147609:1147747) by chromium-webrtc-autoroll · 11 months ago
  29. aa6d4fa Adds WebRTC-DisableRtxRateLimiter for enable/disable RTX rate limiter. by Ying Wang · 11 months ago
  30. f67d1fd OveruseFrameDetector: complete removal of mac rules kill switch. by Markus Handell · 11 months ago
  31. 0613054 Remove unused histograms. by Markus Handell · 11 months ago
  32. 434deda Cleanup RtcpReceiver from using RtcpBandwidthObser callback interface by Danil Chapovalov · 11 months ago
  33. 4858a0d Add test for split-mode SSRC callback by Harald Alvestrand · 11 months ago
  34. 85632b8 Update WebRTC code version (2023-05-23T04:03:48). by webrtc-version-updater · 11 months ago
  35. a8f55c7 Roll chromium_revision 634d3c7e62..fb6508249a (1147498:1147609) by chromium-webrtc-autoroll · 11 months ago
  36. c941cdd Roll chromium_revision 15e9b8d197..634d3c7e62 (1147348:1147498) by chromium-webrtc-autoroll · 11 months ago
  37. 3fea51a Roll chromium_revision 0eaeb41fa6..15e9b8d197 (1147199:1147348) by chromium-webrtc-autoroll · 11 months ago
  38. 13897e6 Change SSRC-passing for MediaChannel from external to callback by Harald Alvestrand · 11 months ago
  39. 5dc4205 Roll chromium_revision 65192f0ef9..0eaeb41fa6 (1147078:1147199) by chromium-webrtc-autoroll · 11 months ago
  40. 1d3452f RequestedResolution - Bug fix by Jonas Oreland · 12 months ago
  41. b7a688c Delete WebRTC.Video.BadCall.* histograms. by Rasmus Brandt · 11 months ago
  42. b401568 Initial copy of flexfec_header_reader_writer. by Yosef Twaik · 12 months ago
  43. 718601a Cleanup RtcpReceiver from passing TransportFeedback via older interface by Danil Chapovalov · 11 months ago
  44. 15feded Increase maximum RTP padding length to 255 bytes by Philipp Hancke · 11 months ago
  45. 194f657 Roll chromium_revision 20c92b363d..65192f0ef9 (1146960:1147078) by chromium-webrtc-autoroll · 11 months ago
  46. 0c85f73 For AV1, disable error resilience on upper temporal layers by Danil Chapovalov · 11 months ago
  47. 3fb338a Update WebRTC code version (2023-05-22T04:03:29). by webrtc-version-updater · 11 months ago
  48. 0483755 Roll chromium_revision f5493a4850..20c92b363d (1146841:1146960) by chromium-webrtc-autoroll · 11 months ago
  49. b1b2c53 Update WebRTC code version (2023-05-21T04:02:48). by webrtc-version-updater · 11 months ago
  50. c9f0b20 Update WebRTC code version (2023-05-20T04:11:18). by webrtc-version-updater · 11 months ago
  51. 0f40079 Roll chromium_revision e7ad7ca1d5..f5493a4850 (1146741:1146841) by chromium-webrtc-autoroll · 11 months ago
  52. 447fc3f Roll chromium_revision f5f7594337..e7ad7ca1d5 (1146453:1146741) by chromium-webrtc-autoroll · 11 months ago
  53. 9a7ca64 Roll chromium_revision d4f384285a..f5f7594337 (1145480:1146453) by chromium-webrtc-autoroll · 11 months ago
  54. cc1ee35 Reland "Avoid recreating VirtualDisplay on format changes." by Linus Nilsson · 11 months ago
  55. 328c514 Reduce precision of RTT in RtrpTransportControllerSend by Danil Chapovalov · 11 months ago
  56. ff75eae Update WebRTC code version (2023-05-18T04:12:20). by webrtc-version-updater · 11 months ago
  57. f6a0680 Roll chromium_revision d59cc17cf9..d4f384285a (1145311:1145480) by chromium-webrtc-autoroll · 11 months ago
  58. 3e39254 Pass rtcp message to RtpTransportController through newer interface by Danil Chapovalov · 11 months ago
  59. a0b1144 Roll chromium_revision 2b0829702f..d59cc17cf9 (1145193:1145311) by chromium-webrtc-autoroll · 11 months ago
  60. 510890b Revert "Avoid recreating VirtualDisplay on format changes." by Mirko Bonadei · 11 months ago
  61. fcd1dfa Avoid recreating VirtualDisplay on format changes. by Linus Nilsson · 11 months ago
  62. 4d0468e Roll chromium_revision 8f747c9bf2..2b0829702f (1144817:1145193) by chromium-webrtc-autoroll · 11 months ago
  63. cb1b73a Update WebRTC code version (2023-05-17T04:12:05). by webrtc-version-updater · 11 months ago
  64. a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 11 months ago
  65. 9a43874 Roll chromium_revision 91c345cf4e..8f747c9bf2 (1144710:1144817) by chromium-webrtc-autoroll · 11 months ago
  66. c37dec2 Set use_cxx to true. by Mirko Bonadei · 12 months ago
  67. ca66eef Roll chromium_revision 1e36b7ebe0..91c345cf4e (1144620:1144710) by chromium-webrtc-autoroll · 11 months ago
  68. 99869ad Roll chromium_revision 30ae698dcc..1e36b7ebe0 (1144471:1144620) by chromium-webrtc-autoroll · 11 months ago
  69. 69bc3e1 Trigger bots by Mirko Bonadei · 11 months ago
  70. f3de65a Change ReceivedFecPacket to have list of ssrcs, seq nums and masks. by Yosef Twaik · 12 months ago
  71. 3a4cfdf Update WebRTC code version (2023-05-16T04:02:28). by webrtc-version-updater · 11 months ago
  72. aaa3b8f Roll chromium_revision 5fb222694e..30ae698dcc (1144315:1144471) by chromium-webrtc-autoroll · 11 months ago
  73. 8cef7c9 Roll chromium_revision 23a25491e5..5fb222694e (1144139:1144315) by chromium-webrtc-autoroll · 11 months ago
  74. 784c339 Expose setCodecPreferences/getCapabilities for android by David Liu · 12 months ago
  75. 54519aa Roll chromium_revision 1f0a8499f0..23a25491e5 (1144024:1144139) by chromium-webrtc-autoroll · 12 months ago
  76. ad363ea Remove spam log from IvfFileWriter. by Mirko Bonadei · 12 months ago
  77. 44ebe2a Refactor HasDataChannels by Tommi · 12 months ago
  78. 3ca0aa5 Set use_cxx to false. by Mirko Bonadei · 12 months ago
  79. 84f6470 Roll chromium_revision bd7ba3acbb..1f0a8499f0 (1143522:1144024) by chromium-webrtc-autoroll · 12 months ago
  80. 487c943 Guard send_codec variable against receive channel access by Harald Alvestrand · 12 months ago
  81. 7924915 Stop decoding video for m-lines which are sendonly or inactive by Philipp Hancke · 12 months ago
  82. 121f1e7 In RtpTransportController reduce information stored about rtcp report blocks by Danil Chapovalov · 12 months ago
  83. eb97ac5 Update WebRTC code version (2023-05-13T04:12:15). by webrtc-version-updater · 12 months ago
  84. 5998ddf Roll chromium_revision 5cdde4d6df..bd7ba3acbb (1143390:1143522) by chromium-webrtc-autoroll · 12 months ago
  85. 7b0d7f4 PipeWire capturer: fix fcntl call when duplicating a file descriptor by Jan Grulich · 12 months ago
  86. 016e57c Roll chromium_revision e89d67b6a3..5cdde4d6df (1143194:1143390) by chromium-webrtc-autoroll · 12 months ago
  87. 32dae4b sdp: accept bundle-only media section without rtcp-mux by Philipp Hancke · 12 months ago
  88. afdc00f Roll chromium_revision f273051fd0..e89d67b6a3 (1142666:1143194) by chromium-webrtc-autoroll · 12 months ago
  89. 63551c6 Initialize RTP modes from callback by Harald Alvestrand · 12 months ago
  90. a09331a Don't write TransmissionOffset when capture time is not set by Andreas Pehrson · 12 months ago
  91. 0280ceb Delete unused member in VideoSendStream by Danil Chapovalov · 12 months ago
  92. 71f80c0 Replace RtcpReceiver::RTT function with RtcpReceiver::AverageRtt with cleaner interface by Danil Chapovalov · 12 months ago
  93. b073283 Roll chromium_revision c8d546c7b8..f273051fd0 (1142541:1142666) by chromium-webrtc-autoroll · 12 months ago
  94. ad7792b rtp_rtcp: fix small typo in enum class RtpPacketMediaType by Alfred E. Heggestad · 12 months ago
  95. 6d0ad4e Update WebRTC code version (2023-05-12T04:01:51). by webrtc-version-updater · 12 months ago
  96. 28e2505f fix some more minor typos by Alfred E. Heggestad · 12 months ago
  97. 36fd351 [Stats] Align RTCStatsMember<T> closer to absl::optional<T>. by Henrik Boström · 12 months ago
  98. 2ec9abd Roll chromium_revision dcb8f1503d..c8d546c7b8 (1142399:1142541) by chromium-webrtc-autoroll · 12 months ago
  99. a79bc6e Roll chromium_revision 6f773ca0c8..dcb8f1503d (1142294:1142399) by chromium-webrtc-autoroll · 12 months ago
  100. efb1292 Remove WEBRTC_EXTERNAL_JSON. by Mirko Bonadei · 12 months ago