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  1. 45d8674 Update WebRTC code version (2023-10-04T04:04:01). by webrtc-version-updater · 5 hours ago main master
  2. 4408575 Reland "Enable SRTP GCM ciphers by default" by Saúl Ibarra Corretgé · 12 hours ago
  3. 8db8824 Fix use-of-uninitialized-value and integer-overflow issues reported by chromium fuzz testing by qwu16 · 20 hours ago lkgr
  4. 40ed3ff Improved event tracing in FrameCadenceAdapter by henrika · 22 hours ago
  5. 9209b50 Update WebRTC code version (2023-10-03T04:04:07). by webrtc-version-updater · 29 hours ago
  6. 3725398 Roll chromium_revision 6a1112fe10..1a86984556 (1204286:1204416) by chromium-webrtc-autoroll · 32 hours ago
  7. 4e5ae32 Roll chromium_revision 45a884046b..6a1112fe10 (1203912:1204286) by chromium-webrtc-autoroll · 36 hours ago
  8. a557468 dcsctp: Only process meaningful FORWARD-TSN by Victor Boivie · 2 days ago
  9. 8e007ba Remove field trial WebRTC-Turn-AllowSystemPorts by Harald Alvestrand · 2 days ago
  10. bce7ce7 Adds support for tracking OnFrame PostTask delta times by henrika · 2 days ago
  11. 4d2d215 Roll chromium_revision cd02ca7e29..45a884046b (1203806:1203912) by chromium-webrtc-autoroll · 2 days ago
  12. 3272a99 Update WebRTC code version (2023-10-02T04:11:24). by webrtc-version-updater · 2 days ago
  13. 316c853 Roll chromium_revision b294093253..cd02ca7e29 (1203697:1203806) by chromium-webrtc-autoroll · 3 days ago
  14. 9cefc24 Update WebRTC code version (2023-10-01T04:06:57). by webrtc-version-updater · 3 days ago
  15. cad0bfa Roll chromium_revision 7be4bb3958..b294093253 (1203597:1203697) by chromium-webrtc-autoroll · 4 days ago
  16. 8212061 Update WebRTC code version (2023-09-30T04:13:02). by webrtc-version-updater · 4 days ago
  17. 7d8ef10 Roll chromium_revision a53f87ab6b..7be4bb3958 (1203480:1203597) by chromium-webrtc-autoroll · 4 days ago
  18. 11f087d Roll chromium_revision 9b3bfd6cf6..a53f87ab6b (1203365:1203480) by chromium-webrtc-autoroll · 4 days ago
  19. 02d5ba1 Roll chromium_revision 3882779f44..9b3bfd6cf6 (1203188:1203365) by chromium-webrtc-autoroll · 5 days ago
  20. dd15070 Adopt EglThread in EglRenderer once again. by Linus Nilsson · 5 days ago
  21. 7d3eb10 Roll chromium_revision 2cd0974cfa..3882779f44 (1202869:1203188) by chromium-webrtc-autoroll · 5 days ago
  22. c27034b Revert "Expose getCapabilities/setCodecPreferences for objc" by Manashi Sarkar · 5 days ago
  23. a2f30e1 Expose getCapabilities/setCodecPreferences for objc by David Liu · 5 days ago
  24. 012c5a3 Remove more Codec-related templating in MediaSession by Philipp Hancke · 5 days ago
  25. 34ec5c3 Clear PacketBuffer on large negative jumps at the start of the video stream by Danil Chapovalov · 5 days ago
  26. e9f9c28 Set permissions on experiments/field_trials.py by Harald Alvestrand · 5 days ago
  27. a61b334 Relax the field trial policy to not require an open bug by Emil Lundmark · 5 days ago
  28. 5e252b7 Register WebRTC-Bwe-SubtractAdditionalBackoffTerm by Emil Lundmark · 5 days ago
  29. 191dd88 Update WebRTC code version (2023-09-29T04:02:17). by webrtc-version-updater · 5 days ago
  30. ea32ab1 Roll chromium_revision 2506ef1631..2cd0974cfa (1202675:1202869) by chromium-webrtc-autoroll · 5 days ago
  31. 5ee308b Roll chromium_revision 116647225e..2506ef1631 (1202523:1202675) by chromium-webrtc-autoroll · 6 days ago
  32. c2bbe4b Revert "Enable SRTP GCM ciphers by default" by Manashi Sarkar · 6 days ago
  33. ebe207f Add field trial for enabling SSL client hello extension permutation by Philipp Hancke · 6 days ago
  34. a1475c2 Roll chromium_revision f7fb707ebb..116647225e (1201480:1202523) by chromium-webrtc-autoroll · 6 days ago
  35. a93f581 dcsctp: Don't generate FORWARD-TSN across stream resets by Victor Boivie · 6 days ago
  36. d863386 Enable SRTP GCM ciphers by default by Saúl Ibarra Corretgé · 6 days ago
  37. 98e71f5 Subtract an additional 5kbps of the bitrate when backing off. by Björn Terelius · 6 days ago
  38. 332c56f MediaSession: ensure transport description factory exists by Philipp Hancke · 6 days ago
  39. bbc7711 Reduce log verbosity in codec selection implementation by Florent Castelli · 6 days ago
  40. 2d508f1 Deprecate old names for EncodedImage::RtpTimestamp accessor and setter by Danil Chapovalov · 6 days ago
  41. f97058e Move static functions in media_session into anonymous namespace by Philipp Hancke · 6 days ago
  42. 21b5ec1 Add AV1 singlecast bitrate limits by Sergey Silkin · 7 days ago
  43. d12e759 Add instructions for adding and removing field trials by Emil Lundmark · 7 days ago
  44. 83894d3 Fire MaybeSignalReadyToSend in a PostTask when recursive by Harald Alvestrand · 7 days ago
  45. 4001cc7 Populate field trial registry by Emil Lundmark · 7 days ago
  46. 5543a96 Add sub-command for validating that field trials conforms to the policy by Emil Lundmark · 7 days ago
  47. 9686a4b Add sub-command for listing expired field trials by Emil Lundmark · 7 days ago
  48. b7fca15 Refactor global variables to be immutable by Emil Lundmark · 7 days ago
  49. 7d81e18 Reformat field_trials.py to follow PEP-8 by Emil Lundmark · 7 days ago
  50. 0505115 Pass the correct abs_capture_timestamp while cloning audio frame by Palak Agarwal · 8 days ago
  51. 3218d74 Roll chromium_revision ae93f006ea..f7fb707ebb (1201360:1201480) by chromium-webrtc-autoroll · 8 days ago
  52. 2bf1b99 Make CreateOffer/CreateAnswer return RTCErrorOr<SessionDescription> by Philipp Hancke · 8 days ago
  53. 06fbe63 dcsctp: Exit deferred stream reset on FORWARD-TSN by Victor Boivie · 8 days ago
  54. 259d95f Roll chromium_revision 3a03dc546a..ae93f006ea (1201238:1201360) by chromium-webrtc-autoroll · 8 days ago
  55. bfc2a35 Remove more codec-related templating by Philipp Hancke · 8 days ago
  56. 7829daf Update WebRTC code version (2023-09-26T04:02:19). by webrtc-version-updater · 8 days ago
  57. 78c119c Remove check on last_packet_received_time_ as it's no longer used. by Ying Wang · 8 days ago
  58. 9b5d795 Roll chromium_revision eb86ccf4cd..3a03dc546a (1201083:1201238) by chromium-webrtc-autoroll · 8 days ago
  59. 0ca4d62 Roll chromium_revision ae69785833..eb86ccf4cd (1200919:1201083) by chromium-webrtc-autoroll · 9 days ago
  60. 77df7ff dcsctp: Cleanup use of matchers by Victor Boivie · 9 days ago
  61. 7892f05 Configure Pylint to follow PEP-8 by Emil Lundmark · 9 days ago
  62. 7d1aff6 Unify RTP payload type validity checking by Philipp Hancke · 9 days ago
  63. 6bf2d31 Change PortInterface::Type to string_view and make type_ member const by Tommi · 9 days ago
  64. 070d386 Roll chromium_revision 286dbc6af0..ae69785833 (1200141:1200919) by chromium-webrtc-autoroll · 9 days ago
  65. 29d4a01 Reland: use loss based bwe v2 in the start phase. by Diep Bui · 9 days ago
  66. ba97eec Add string_view overload for Wrap method by Artem Titov · 9 days ago
  67. b4d4bbc Revert "Clean up last_packet_received_time_ as it's no longer used." by Björn Terelius · 9 days ago
  68. 9c58483 Rename EncodedImage property Timetamp to RtpTimestamp by Danil Chapovalov · 10 days ago
  69. bbf27e0 Remove NSApplicationActivateIgnoringOtherApps by Johannes Kron · 10 days ago
  70. 850296b Reapply "dcsctp: Negotiate zero checksum" by Victor Boivie · 11 days ago
  71. 63c50f5 Update WebRTC code version (2023-09-23T04:12:34). by webrtc-version-updater · 11 days ago
  72. 2f4bc64 Clean up last_packet_received_time_ as it's no longer used. by Ying Wang · 11 days ago
  73. d2f4cf9 Roll chromium_revision c066d24408..286dbc6af0 (1200027:1200141) by chromium-webrtc-autoroll · 11 days ago
  74. 4aa2b40 Revert "Use loss based bwe v2 in the start phase." by Diep Bui · 11 days ago
  75. b6c7ddd Use loss based bwe v2 in the start phase. by Diep Bui · 12 days ago
  76. b6ea0b2 Direcly call configure_reclient_cfgs.py instead of indirectly via fetch_reclient_cfgs.py by Björn Terelius · 12 days ago
  77. 1db3980 Remove upper_link_capacity from loss_based_bwe_v2. by Diep Bui · 12 days ago
  78. 70eec6d Configure YAPF to follow PEP-8 altogether by Emil Lundmark · 12 days ago
  79. 4b39e86 Update WebRTC code version (2023-09-22T04:11:01). by webrtc-version-updater · 12 days ago
  80. 8500974 Roll chromium_revision 4b9a788892..c066d24408 (1199866:1200027) by chromium-webrtc-autoroll · 12 days ago
  81. 047eeb4 Roll chromium_revision f473cfebae..4b9a788892 (1199635:1199866) by chromium-webrtc-autoroll · 12 days ago
  82. e3e030e Roll chromium_revision 54d127d7c9..f473cfebae (1199499:1199635) by chromium-webrtc-autoroll · 13 days ago
  83. e887cbe Roll chromium_revision b3921f4990..54d127d7c9 (1198700:1199499) by chromium-webrtc-autoroll · 13 days ago
  84. 7ee64bd Remove the upper link capacity usage in the loss based bwe. by Diep Bui · 13 days ago
  85. c951d1b audio: fix some typos by Alfred E. Heggestad · 13 days ago
  86. 6fc4d97 Make WEBRTC_UNSAFE_FUZZER_MODE dependent only on use_fuzzing_engine by Greg Thompson · 14 days ago
  87. 46da472 Revert "mac: Work around an inccorect availability annotation in the 13.3 SDK" by Avi Drissman · 14 days ago
  88. 5551776 Reject attempts to change the media kind for a m-line with a previously used mid by Philipp Hancke · 14 days ago
  89. ec82627 Look through all candidates before falling back to default packetization by Emil Lundmark · 14 days ago
  90. f14dfed Move codecs() to MediaContentDescription by Philipp Hancke · 2 weeks ago
  91. ae82df7 Add codec name H265 to support H265 in WebRTC by qwu16 · 2 weeks ago
  92. 9596002 Update WebRTC code version (2023-09-20T04:02:40). by webrtc-version-updater · 2 weeks ago
  93. dacd1fa Roll chromium_revision b63463aa70..b3921f4990 (1198567:1198700) by chromium-webrtc-autoroll · 2 weeks ago
  94. 7917525 Roll chromium_revision dfc3d16403..b63463aa70 (1198356:1198567) by chromium-webrtc-autoroll · 2 weeks ago
  95. 4b583c7 Roll chromium_revision eef62e8a0c..dfc3d16403 (1197906:1198356) by chromium-webrtc-autoroll · 2 weeks ago
  96. f8c70c9 fix: Handle out-of-range device index after GetDevicesInfo by Youfa · 2 weeks ago
  97. 2e7ed0d Roll chromium_revision 6ac7929166..eef62e8a0c (1190797:1197906) by Jeremy Leconte · 2 weeks ago
  98. e14d122 Remove deprecated SendRtp and SendRtcp functions by Harald Alvestrand · 2 weeks ago
  99. 090699a Delete deprecated RtpSource timestamp_ms constructor and accessor by Danil Chapovalov · 2 weeks ago
  100. 7cdf66f Add local capture clock offset to video RtpPacketInfos by Olov Brändström · 2 weeks ago