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webrtc / src / / 4c1e9598a30b46500ef1b145e3e9965fc34913e3
  1. 4c1e959 Change flexfec header reader to parse according to updated RFC. by Yosef Twaik · 1 year, 11 months ago
  2. e4a9a6d Update WebRTC code version (2023-05-30T04:02:06). by webrtc-version-updater · 1 year, 11 months ago
  3. c5e4bcc Roll chromium_revision 599c746c73..c90a8a46d7 (1150194:1150306) by chromium-webrtc-autoroll · 1 year, 11 months ago
  4. 4b14cb7 Roll chromium_revision fa2e063162..599c746c73 (1150086:1150194) by chromium-webrtc-autoroll · 1 year, 11 months ago
  5. 4aaacb4 Update WebRTC code version (2023-05-29T04:03:50). by webrtc-version-updater · 1 year, 11 months ago
  6. e641a97 In RtcpReceiver remove redundand way to represent RTCP report blocks by Danil Chapovalov · 1 year, 11 months ago
  7. b9de471 Update WebRTC code version (2023-05-28T04:11:22). by webrtc-version-updater · 1 year, 11 months ago
  8. 98185b9 Roll chromium_revision 99b12997bf..fa2e063162 (1150050:1150086) by chromium-webrtc-autoroll · 1 year, 11 months ago
  9. a294353 Use type raw for video_codec_perf_tests. by Mirko Bonadei · 1 year, 11 months ago
  10. 01c2efc Roll chromium_revision bddf6cbe18..99b12997bf (1149812:1150050) by chromium-webrtc-autoroll · 1 year, 11 months ago
  11. 9bc8d05 Update WebRTC code version (2023-05-27T04:12:09). by webrtc-version-updater · 1 year, 11 months ago
  12. 9ac543c Roll chromium_revision 1fc947a5da..bddf6cbe18 (1149703:1149812) by chromium-webrtc-autoroll · 1 year, 11 months ago
  13. 87e74f9 Remove unused combined_audio_video_bwe. by Yury Yarashevich · 1 year, 11 months ago
  14. 2bb686d Stop running low_bandwith_audio_tests. by Jeremy Leconte · 1 year, 11 months ago
  15. 6490999 Roll chromium_revision aae661725b..1fc947a5da (1148994:1149703) by chromium-webrtc-autoroll · 1 year, 11 months ago
  16. f0820ff Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay by Rasmus Brandt · 1 year, 11 months ago
  17. 9caef2a Use a constant for invalid PipeWire file descriptor by Jan Grulich · 1 year, 11 months ago
  18. 0f1a2c5 Change StreamDataCounters to use Timestamp instead of int64_t by Danil Chapovalov · 1 year, 11 months ago
  19. 5f32fa4 Delete MediaBaseChannel class by Harald Alvestrand · 1 year, 11 months ago
  20. 4f1dcbb doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c by Li-Yu Yu · 1 year, 11 months ago
  21. f53b343 Cleanup RtcpTransceiver dependency on webrtc::Transport by Danil Chapovalov · 1 year, 11 months ago
  22. 5f38949 Allow single-mline offers without BUNDLE group when using max-bundle by Philipp Hancke · 1 year, 11 months ago
  23. dff6e25 Update WebRTC code version (2023-05-26T04:05:22). by webrtc-version-updater · 1 year, 11 months ago
  24. 6e23fa5 Cleanup WebRTC-PayloadTypes-Lower-Dynamic-Range trial by Philipp Hancke · 1 year, 11 months ago
  25. 56d1260 PipeWire video capture: split portal and PipeWire implementations by Jan Grulich · 1 year, 11 months ago
  26. 2264e7a Fixes distortion in WGC screen capture path by henrika · 1 year, 11 months ago
  27. 40a0fa9 Add new padding mode to RtpPacketHistory by Per K · 1 year, 11 months ago
  28. 4206d31 [Analysis] Add new thresholds config schema by Beining Chen · 1 year, 11 months ago
  29. cfd4cd0 Introduce AddDefaultRecvStreamForTesting to VideoReceiveChannel API by Harald Alvestrand · 1 year, 11 months ago
  30. 5c35d08 Replace "RTRR" with "RRTR" by Philipp Hancke · 1 year, 11 months ago
  31. 1ecac13 Roll chromium_revision be3e47cd99..aae661725b (1148555:1148994) by chromium-webrtc-autoroll · 1 year, 11 months ago
  32. f4d0a49 Adopt EglThread in EglRenderer by Linus Nilsson · 1 year, 11 months ago
  33. 1cc41ea Remove unused Win32Window class by Philipp Hancke · 1 year, 11 months ago
  34. 0f13765 Delete RTC[NonStandard/Restricted]StatsMember. by Henrik Boström · 1 year, 11 months ago
  35. 621cb29 Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. by Rasmus Brandt · 1 year, 11 months ago
  36. 8ac66a2 Update WebRTC code version (2023-05-25T04:12:48). by webrtc-version-updater · 1 year, 11 months ago
  37. ec863a2 Roll chromium_revision 8f46ad499d..be3e47cd99 (1148441:1148555) by chromium-webrtc-autoroll · 1 year, 11 months ago
  38. 979b047 Revert "Temporarily add dummy trackId to unblock roll." by Henrik Boström · 1 year, 11 months ago
  39. aa1ad7d In RtcpTransciever refactor outgoing transport interface by Danil Chapovalov · 1 year, 11 months ago
  40. f4d762e docs: explain release note process by Philipp Hancke · 1 year, 11 months ago
  41. 2057d71 [Stats] Delete unused NonStandardGroupId. by Henrik Boström · 1 year, 11 months ago
  42. 61bacd1 Enable WebRTC-SplitMediaChannel by default by Harald Alvestrand · 1 year, 11 months ago
  43. 3df4178 Temporarily add dummy trackId to unblock roll. by Henrik Boström · 1 year, 11 months ago
  44. 4e231ee Delete deprecated 'track' and 'stream' metrics from WebRTC. by Henrik Boström · 1 year, 11 months ago
  45. 54c37a5 Roll chromium_revision 9378e3160b..8f46ad499d (1148314:1148441) by chromium-webrtc-autoroll · 1 year, 11 months ago
  46. 18898d7 Update WebRTC code version (2023-05-24T04:17:39). by webrtc-version-updater · 1 year, 11 months ago
  47. 1920430 Roll chromium_revision 3d8a0c6a22..9378e3160b (1148203:1148314) by chromium-webrtc-autoroll · 1 year, 11 months ago
  48. 531383c Roll chromium_revision bbf4ff3290..3d8a0c6a22 (1148035:1148203) by chromium-webrtc-autoroll · 1 year, 11 months ago
  49. 4daa4e6 Roll chromium_revision c025c4ac7b..bbf4ff3290 (1147854:1148035) by chromium-webrtc-autoroll · 1 year, 11 months ago
  50. a7d1081 Revert "pipewire capturer: Reduce the amount of copying" by Alexander Cooper · 1 year, 11 months ago
  51. 2eacbbc Roll chromium_revision 692840e030..c025c4ac7b (1147747:1147854) by chromium-webrtc-autoroll · 1 year, 11 months ago
  52. 33697291 Add EglThread class wrapping EglConnection and handler. by Linus Nilsson · 1 year, 11 months ago
  53. ff35a37 Unit tests for MediaChannel creation API by Harald Alvestrand · 1 year, 11 months ago
  54. 98f47a3 Delete redundant member StreamDataCounters::last_packet_received_time by Danil Chapovalov · 1 year, 11 months ago
  55. 0328190 Add video_codec_perf_tests to desktop and android perf test suites by Sergey Silkin · 1 year, 11 months ago
  56. e3441ec Roll chromium_revision fb6508249a..692840e030 (1147609:1147747) by chromium-webrtc-autoroll · 1 year, 11 months ago
  57. aa6d4fa Adds WebRTC-DisableRtxRateLimiter for enable/disable RTX rate limiter. by Ying Wang · 1 year, 11 months ago
  58. f67d1fd OveruseFrameDetector: complete removal of mac rules kill switch. by Markus Handell · 1 year, 11 months ago
  59. 0613054 Remove unused histograms. by Markus Handell · 1 year, 11 months ago
  60. 434deda Cleanup RtcpReceiver from using RtcpBandwidthObser callback interface by Danil Chapovalov · 1 year, 11 months ago
  61. 4858a0d Add test for split-mode SSRC callback by Harald Alvestrand · 1 year, 11 months ago
  62. 85632b8 Update WebRTC code version (2023-05-23T04:03:48). by webrtc-version-updater · 1 year, 11 months ago
  63. a8f55c7 Roll chromium_revision 634d3c7e62..fb6508249a (1147498:1147609) by chromium-webrtc-autoroll · 1 year, 11 months ago
  64. c941cdd Roll chromium_revision 15e9b8d197..634d3c7e62 (1147348:1147498) by chromium-webrtc-autoroll · 1 year, 11 months ago
  65. 3fea51a Roll chromium_revision 0eaeb41fa6..15e9b8d197 (1147199:1147348) by chromium-webrtc-autoroll · 1 year, 11 months ago
  66. 13897e6 Change SSRC-passing for MediaChannel from external to callback by Harald Alvestrand · 1 year, 11 months ago
  67. 5dc4205 Roll chromium_revision 65192f0ef9..0eaeb41fa6 (1147078:1147199) by chromium-webrtc-autoroll · 1 year, 11 months ago
  68. 1d3452f RequestedResolution - Bug fix by Jonas Oreland · 1 year, 11 months ago
  69. b7a688c Delete WebRTC.Video.BadCall.* histograms. by Rasmus Brandt · 1 year, 11 months ago
  70. b401568 Initial copy of flexfec_header_reader_writer. by Yosef Twaik · 1 year, 11 months ago
  71. 718601a Cleanup RtcpReceiver from passing TransportFeedback via older interface by Danil Chapovalov · 1 year, 11 months ago
  72. 15feded Increase maximum RTP padding length to 255 bytes by Philipp Hancke · 1 year, 11 months ago
  73. 194f657 Roll chromium_revision 20c92b363d..65192f0ef9 (1146960:1147078) by chromium-webrtc-autoroll · 1 year, 11 months ago
  74. 0c85f73 For AV1, disable error resilience on upper temporal layers by Danil Chapovalov · 1 year, 11 months ago
  75. 3fb338a Update WebRTC code version (2023-05-22T04:03:29). by webrtc-version-updater · 1 year, 11 months ago
  76. 0483755 Roll chromium_revision f5493a4850..20c92b363d (1146841:1146960) by chromium-webrtc-autoroll · 1 year, 11 months ago
  77. b1b2c53 Update WebRTC code version (2023-05-21T04:02:48). by webrtc-version-updater · 1 year, 11 months ago
  78. c9f0b20 Update WebRTC code version (2023-05-20T04:11:18). by webrtc-version-updater · 1 year, 11 months ago
  79. 0f40079 Roll chromium_revision e7ad7ca1d5..f5493a4850 (1146741:1146841) by chromium-webrtc-autoroll · 1 year, 11 months ago
  80. 447fc3f Roll chromium_revision f5f7594337..e7ad7ca1d5 (1146453:1146741) by chromium-webrtc-autoroll · 1 year, 11 months ago
  81. 9a7ca64 Roll chromium_revision d4f384285a..f5f7594337 (1145480:1146453) by chromium-webrtc-autoroll · 1 year, 11 months ago
  82. cc1ee35 Reland "Avoid recreating VirtualDisplay on format changes." by Linus Nilsson · 1 year, 11 months ago
  83. 328c514 Reduce precision of RTT in RtrpTransportControllerSend by Danil Chapovalov · 1 year, 11 months ago
  84. ff75eae Update WebRTC code version (2023-05-18T04:12:20). by webrtc-version-updater · 1 year, 11 months ago
  85. f6a0680 Roll chromium_revision d59cc17cf9..d4f384285a (1145311:1145480) by chromium-webrtc-autoroll · 1 year, 11 months ago
  86. 3e39254 Pass rtcp message to RtpTransportController through newer interface by Danil Chapovalov · 1 year, 11 months ago
  87. a0b1144 Roll chromium_revision 2b0829702f..d59cc17cf9 (1145193:1145311) by chromium-webrtc-autoroll · 1 year, 11 months ago
  88. 510890b Revert "Avoid recreating VirtualDisplay on format changes." by Mirko Bonadei · 1 year, 11 months ago
  89. fcd1dfa Avoid recreating VirtualDisplay on format changes. by Linus Nilsson · 1 year, 11 months ago
  90. 4d0468e Roll chromium_revision 8f747c9bf2..2b0829702f (1144817:1145193) by chromium-webrtc-autoroll · 1 year, 11 months ago
  91. cb1b73a Update WebRTC code version (2023-05-17T04:12:05). by webrtc-version-updater · 1 year, 11 months ago
  92. a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 1 year, 11 months ago
  93. 9a43874 Roll chromium_revision 91c345cf4e..8f747c9bf2 (1144710:1144817) by chromium-webrtc-autoroll · 1 year, 11 months ago
  94. c37dec2 Set use_cxx to true. by Mirko Bonadei · 1 year, 11 months ago
  95. ca66eef Roll chromium_revision 1e36b7ebe0..91c345cf4e (1144620:1144710) by chromium-webrtc-autoroll · 1 year, 11 months ago
  96. 99869ad Roll chromium_revision 30ae698dcc..1e36b7ebe0 (1144471:1144620) by chromium-webrtc-autoroll · 1 year, 11 months ago
  97. 69bc3e1 Trigger bots by Mirko Bonadei · 1 year, 11 months ago
  98. f3de65a Change ReceivedFecPacket to have list of ssrcs, seq nums and masks. by Yosef Twaik · 1 year, 11 months ago
  99. 3a4cfdf Update WebRTC code version (2023-05-16T04:02:28). by webrtc-version-updater · 1 year, 11 months ago
  100. aaa3b8f Roll chromium_revision 5fb222694e..30ae698dcc (1144315:1144471) by chromium-webrtc-autoroll · 1 year, 11 months ago