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  1. cc9dcaf Roll chromium_revision 2d895ae064..fb6772e9b1 (1297992:1298140) by chromium-webrtc-autoroll · 27 hours ago main master
  2. 53d43d6 SimulatedNetworkNode::Builder use BuiltInNetworkBehaviorConfig.link_capacity by Per K · 27 hours ago lkgr
  3. f42d2b9 Include-what-you-use pc/media_session by Harald Alvestrand · 36 hours ago
  4. 26a082c Introduce a mode that lets NetworkEmulationManager ignore DTLS handshake sizes. by Sergey Sukhanov · 31 hours ago
  5. f20ed3e Add option to provide Environment for CongestionConroller construction by Danil Chapovalov · 32 hours ago
  6. 5f3fc50 Split CreateEncodingSettings by Sergey Silkin · 34 hours ago
  7. 35af7cc Move sharding logic to the correct context by Christoffer Dewerin · 33 hours ago
  8. 1545cdc Roll chromium_revision 67ad773ea4..2d895ae064 (1297526:1297992) by chromium-webrtc-autoroll · 35 hours ago
  9. 94dfe1c Fix NetworkMonitor race condition when dispatching native observers by Guy Hershenbaum · 3 months ago
  10. fc57037 Revert "Split digest methods from ssl target into digest target" by Mirko Bonadei · 2 days ago
  11. 8de2cf7 Update WebRTC code version (2024-05-08T04:01:40). by webrtc-version-updater · 2 days ago
  12. 0d037df Reland "Add more accurate support for changing capacity in SimulatedNetwork" by Per K · 2 days ago
  13. cc92b6e Add ProbeController::EnableRepeatedInitialProbing by Per K · 2 days ago
  14. ae8cf2b Roll chromium_revision 2c3468559e..67ad773ea4 (1297387:1297526) by chromium-webrtc-autoroll · 2 days ago
  15. f126e8a Delete implicit layering mode. by Sergey Silkin · 2 days ago
  16. 47bfe39 Split digest methods from ssl target into digest target by Philipp Hancke · 2 days ago
  17. 2561871 Make perkj owner of call/ by Per K · 2 days ago
  18. 141f4c1 Provide mechanism to make codec decisions per-transceiver by Harald Alvestrand · 2 days ago
  19. 30e3553 Roll chromium_revision 5faeabaa35..2c3468559e (1297280:1297387) by chromium-webrtc-autoroll · 2 days ago
  20. 6866da1 Revert "Add more accurate support for changing capacity in SimulatedNetwork" by Per Kjellander · 2 days ago
  21. fba7d84 Refactor event log analyser bindings unit test by Björn Terelius · 2 days ago
  22. 51a70c0 Add more accurate support for changing capacity in SimulatedNetwork by Per K · 3 days ago
  23. 2ee83c1 Provide Environment for ReceiveSideConfestionController construction by Danil Chapovalov · 3 days ago
  24. f5e9f11 Delete WebRTC-LibaomAv1Encoder-DisableFrameDropping by Sergey Silkin · 3 days ago
  25. da648b5 Update WebRTC code version (2024-05-07T04:06:18). by webrtc-version-updater · 3 days ago
  26. 94f1d9f Roll chromium_revision d2acc67c32..5faeabaa35 (1297067:1297280) by chromium-webrtc-autoroll · 3 days ago
  27. 84e9055 Roll chromium_revision 7f4bb5d096..d2acc67c32 (1296883:1297067) by chromium-webrtc-autoroll · 3 days ago
  28. 3f10f65 sdp: answer with spec msid when msid support is unknown by Philipp Hancke · 3 weeks ago
  29. 1e5f88c Make muted param in GetAudio optional. by Jakob Ivarsson · 3 days ago
  30. 86cd7a3 Roll chromium_revision 27cbc72c1a..7f4bb5d096 (1296748:1296883) by chromium-webrtc-autoroll · 3 days ago
  31. 89679bf sdp: document existing mid backfill cornercases by Philipp Hancke · 6 days ago
  32. 01ff41e Cleanup expired field trial WebRTC-Avx2SupportKillSwitch by Danil Chapovalov · 3 days ago
  33. 9559b2d Roll chromium_revision d6f2f1ce6b..27cbc72c1a (1296010:1296748) by chromium-webrtc-autoroll · 3 days ago
  34. 549c413 Roll chromium_revision 669d8ffcd7..d6f2f1ce6b (1292426:1296010) by Mirko Bonadei · 3 days ago
  35. 5dbc4a4 Temporary disable sharding on Fuchsia bots. by Jeremy Leconte · 4 days ago
  36. 53156f0 Update WebRTC code version (2024-05-06T04:02:48). by webrtc-version-updater · 4 days ago
  37. a2e33ed Update WebRTC code version (2024-05-05T04:01:32). by webrtc-version-updater · 5 days ago
  38. 00670e7 Update WebRTC code version (2024-05-04T04:05:48). by webrtc-version-updater · 6 days ago
  39. 853e247 Set full path to input video in EncodeDecode test by Sergey Silkin · 6 days ago
  40. 8b7d89a Cleanup expired field trial WebRTC-Video-QualityRampupSettings by Danil Chapovalov · 6 days ago
  41. 5ed460a Remove WebRTC-BoostedScreenshareQp by Sergey Silkin · 6 days ago
  42. 8a5f807 Reland "h264: bail out early when failing to parse SPS/PPS ids" by Danil Chapovalov · 6 days ago
  43. b1a71aa Introduce GCS dependencies support in DEPS autoroller by Byoungchan Lee · 6 days ago
  44. 605d00b VideoFrameBuffer: remove TODO. by Markus Handell · 6 days ago
  45. 111d957 Cleanup unused field trial WebRTC-Video-BandwidthQualityScalerSettings by Danil Chapovalov · 6 days ago
  46. 5b64329 Use proper TRACE_EVENT_ASYNC_STEP macro with perfetto by Evan Shrubsole · 7 days ago
  47. 8410b6e Add --screencast and --frame_drop flags to EncodeDecode test by Sergey Silkin · 9 days ago
  48. e1607ed Revert "h264: bail out early when failing to parse SPS/PPS ids" by Mirko Bonadei · 6 days ago
  49. 6982188 Update WebRTC code version (2024-05-03T04:04:17). by webrtc-version-updater · 7 days ago
  50. 363917a Add support for receiving CongestionControlFeedback to RTCPReceiver by Per K · 7 days ago
  51. 1a436f7 Remove AudioFrameOperations::Add, ApplyHalfGain and Scale. by Tommi · 9 days ago
  52. 81eca83 Revert "Remove unused WebRTC-Bwe-InjectedCongestionController" by Qingsi Wang · 9 days ago
  53. 62735dd In Vp9 encoder references fuzzer ignore EncoderInfoOverride field trial by Danil Chapovalov · 7 days ago
  54. 4344eb7 h264: bail out early when failing to parse SPS/PPS ids by Philipp Hancke · 10 days ago
  55. d48a18f Limit pacingfactor by upper link capacity estimate. by Per K · 7 days ago
  56. fa87037 Always use Perfetto when build_with_chromium by Evan Shrubsole · 7 days ago
  57. 55f6613 Retry initial probe if it times out and BWE has not been updated. by Per K · 7 days ago
  58. eeff850 Adding the option to experiment with the max_allowed_excess_render_blocks parameter. by Jesús de Vicente Peña · 9 days ago
  59. 3baefbf Return absl::optional<size_t> from FileWrapper::FileSize() by Björn Terelius · 2 weeks ago
  60. af65d4b Update WebRTC code version (2024-05-02T04:06:36). by webrtc-version-updater · 8 days ago
  61. 57b09ec Update AudioFrameOperations to require ArrayView by Tommi · 9 days ago
  62. acfd279 av1: make packetization generate more evenly sized packets by Philipp Hancke · 9 days ago
  63. 1f36798 Start using ArrayView in AudioFrame, update PushResampler by Tommi · 9 days ago
  64. 652bd28 Query EncoderInfoSettings through propagated field trials by Danil Chapovalov · 2 weeks ago
  65. a345880 Add IWYU export pragmas to gtest/gmock by Evan Shrubsole · 9 days ago
  66. b2b6166 Make AudioFrame::channel_layout_ private and check for valid values by Tommi · 10 days ago
  67. 1ce9a17 Generate privacy manifest when creating Apple Framework by Byoungchan Lee · 9 days ago
  68. cd09858 Convert decoder TRACE_EVENT to flows by Evan Shrubsole · 9 days ago
  69. c3cdab0 Update WebRTC code version (2024-04-30T04:14:10). by webrtc-version-updater · 10 days ago
  70. ffb49c2 Add Monorail -> Google Issue Tracker map. by Mirko Bonadei · 10 days ago
  71. d78e30e Deprecate cricket::VideoCodec and cricket::AudioCodec by Harald Alvestrand · 10 days ago
  72. 64437e8 Calculate the audio level of audio packets before encoded transforms by Tony Herre · 10 days ago
  73. 047238e WebRTC perfetto chromium integration by Evan Shrubsole · 10 days ago
  74. 569849e Move call/simulated_network to test/network by Per K · 10 days ago
  75. c21a150 Use Google issue tracker bug IDs in the field trial registry by Emil Lundmark · 13 days ago
  76. 6ab9085 Fix iwyu error introduced recently. by Tommi · 11 days ago
  77. 3e7a550 Update WebRTC code version (2024-04-29T04:02:07). by webrtc-version-updater · 11 days ago
  78. 7e41c06 Deprecate the StreamInterface::SignalEvent sigslot by Tommi · 11 days ago
  79. e92f409 Update WebRTC code version (2024-04-28T04:02:16). by webrtc-version-updater · 12 days ago
  80. c75ee61 Update WebRTC code version (2024-04-27T04:07:22). by webrtc-version-updater · 13 days ago
  81. 5ccd44b Remove EncodedData::reference_buffers. by philipel · 13 days ago
  82. 3703b35 Using Ntp times for the absolute send time. by Jesús de Vicente Peña · 13 days ago
  83. a130e37 Reland "lets try again" by Christoffer Dewerin · 13 days ago
  84. cfddbfe Revert "lets try again" by Christoffer Dewerin · 13 days ago
  85. f03b06e lets try again by Christoffer Dewerin · 13 days ago
  86. 0d9e83c testing by Christoffer Dewerin · 13 days ago
  87. b386d47 Update WebRTC code version (2024-04-26T04:03:31). by webrtc-version-updater · 14 days ago
  88. decc48f Fix 'Screen flickering on ScreenCapturerWinDirectx' by memetao · 5 months ago
  89. 3772354 Roll chromium_revision ddd32f326f..669d8ffcd7 (1292311:1292426) by chromium-webrtc-autoroll · 2 weeks ago
  90. b5f2442 dcsctp: Remove dead code by Victor Boivie · 2 weeks ago
  91. 2e1a2cd Make stats analysis working with empty layers (bitrate=0) by Sergey Silkin · 2 weeks ago
  92. d009421 Roll chromium_revision 8b3f58c31e..ddd32f326f (1292052:1292311) by chromium-webrtc-autoroll · 2 weeks ago
  93. b85b4c0 Reland "New video encoder API." by philipel · 2 weeks ago
  94. b0e7057 Introduce the TransformerHost interface by Harald Alvestrand · 2 weeks ago
  95. 28d07dd dcsctp: Compute RTO with higher precision by Victor Boivie · 2 weeks ago
  96. 1a3120f Move some integration test functions to the .cc file by Harald Alvestrand · 2 weeks ago
  97. f9a5ed0 Update WebRTC code version (2024-04-25T04:03:46). by webrtc-version-updater · 2 weeks ago
  98. caa1201 Roll chromium_revision f24efc069c..8b3f58c31e (1291744:1292052) by chromium-webrtc-autoroll · 2 weeks ago
  99. db50b03 Add perfetto build config by Evan Shrubsole · 2 weeks ago
  100. 2a66531 Delete deprecated CreateVideoEncoderSoftwareFallbackWrapper by Danil Chapovalov · 3 weeks ago