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webrtc / src / 01ff41e594af0b40dd9c502f0d506a2124f662ac
  1. 01ff41e Cleanup expired field trial WebRTC-Avx2SupportKillSwitch by Danil Chapovalov · 10 months ago
  2. 9559b2d Roll chromium_revision d6f2f1ce6b..27cbc72c1a (1296010:1296748) by chromium-webrtc-autoroll · 10 months ago
  3. 549c413 Roll chromium_revision 669d8ffcd7..d6f2f1ce6b (1292426:1296010) by Mirko Bonadei · 10 months ago
  4. 5dbc4a4 Temporary disable sharding on Fuchsia bots. by Jeremy Leconte · 10 months ago
  5. 53156f0 Update WebRTC code version (2024-05-06T04:02:48). by webrtc-version-updater · 10 months ago
  6. a2e33ed Update WebRTC code version (2024-05-05T04:01:32). by webrtc-version-updater · 10 months ago
  7. 00670e7 Update WebRTC code version (2024-05-04T04:05:48). by webrtc-version-updater · 10 months ago
  8. 853e247 Set full path to input video in EncodeDecode test by Sergey Silkin · 10 months ago
  9. 8b7d89a Cleanup expired field trial WebRTC-Video-QualityRampupSettings by Danil Chapovalov · 10 months ago
  10. 5ed460a Remove WebRTC-BoostedScreenshareQp by Sergey Silkin · 10 months ago
  11. 8a5f807 Reland "h264: bail out early when failing to parse SPS/PPS ids" by Danil Chapovalov · 10 months ago
  12. b1a71aa Introduce GCS dependencies support in DEPS autoroller by Byoungchan Lee · 10 months ago
  13. 605d00b VideoFrameBuffer: remove TODO. by Markus Handell · 10 months ago
  14. 111d957 Cleanup unused field trial WebRTC-Video-BandwidthQualityScalerSettings by Danil Chapovalov · 10 months ago
  15. 5b64329 Use proper TRACE_EVENT_ASYNC_STEP macro with perfetto by Evan Shrubsole · 10 months ago
  16. 8410b6e Add --screencast and --frame_drop flags to EncodeDecode test by Sergey Silkin · 10 months ago
  17. e1607ed Revert "h264: bail out early when failing to parse SPS/PPS ids" by Mirko Bonadei · 10 months ago
  18. 6982188 Update WebRTC code version (2024-05-03T04:04:17). by webrtc-version-updater · 10 months ago
  19. 363917a Add support for receiving CongestionControlFeedback to RTCPReceiver by Per K · 10 months ago
  20. 1a436f7 Remove AudioFrameOperations::Add, ApplyHalfGain and Scale. by Tommi · 10 months ago
  21. 81eca83 Revert "Remove unused WebRTC-Bwe-InjectedCongestionController" by Qingsi Wang · 10 months ago
  22. 62735dd In Vp9 encoder references fuzzer ignore EncoderInfoOverride field trial by Danil Chapovalov · 10 months ago
  23. 4344eb7 h264: bail out early when failing to parse SPS/PPS ids by Philipp Hancke · 11 months ago
  24. d48a18f Limit pacingfactor by upper link capacity estimate. by Per K · 10 months ago
  25. fa87037 Always use Perfetto when build_with_chromium by Evan Shrubsole · 10 months ago
  26. 55f6613 Retry initial probe if it times out and BWE has not been updated. by Per K · 10 months ago
  27. eeff850 Adding the option to experiment with the max_allowed_excess_render_blocks parameter. by Jesús de Vicente Peña · 10 months ago
  28. 3baefbf Return absl::optional<size_t> from FileWrapper::FileSize() by Björn Terelius · 11 months ago
  29. af65d4b Update WebRTC code version (2024-05-02T04:06:36). by webrtc-version-updater · 10 months ago
  30. 57b09ec Update AudioFrameOperations to require ArrayView by Tommi · 10 months ago
  31. acfd279 av1: make packetization generate more evenly sized packets by Philipp Hancke · 10 months ago
  32. 1f36798 Start using ArrayView in AudioFrame, update PushResampler by Tommi · 10 months ago
  33. 652bd28 Query EncoderInfoSettings through propagated field trials by Danil Chapovalov · 11 months ago
  34. a345880 Add IWYU export pragmas to gtest/gmock by Evan Shrubsole · 10 months ago
  35. b2b6166 Make AudioFrame::channel_layout_ private and check for valid values by Tommi · 10 months ago
  36. 1ce9a17 Generate privacy manifest when creating Apple Framework by Byoungchan Lee · 10 months ago
  37. cd09858 Convert decoder TRACE_EVENT to flows by Evan Shrubsole · 10 months ago
  38. c3cdab0 Update WebRTC code version (2024-04-30T04:14:10). by webrtc-version-updater · 10 months ago
  39. ffb49c2 Add Monorail -> Google Issue Tracker map. by Mirko Bonadei · 11 months ago
  40. d78e30e Deprecate cricket::VideoCodec and cricket::AudioCodec by Harald Alvestrand · 11 months ago
  41. 64437e8 Calculate the audio level of audio packets before encoded transforms by Tony Herre · 11 months ago
  42. 047238e WebRTC perfetto chromium integration by Evan Shrubsole · 11 months ago
  43. 569849e Move call/simulated_network to test/network by Per K · 11 months ago
  44. c21a150 Use Google issue tracker bug IDs in the field trial registry by Emil Lundmark · 11 months ago
  45. 6ab9085 Fix iwyu error introduced recently. by Tommi · 11 months ago
  46. 3e7a550 Update WebRTC code version (2024-04-29T04:02:07). by webrtc-version-updater · 11 months ago
  47. 7e41c06 Deprecate the StreamInterface::SignalEvent sigslot by Tommi · 11 months ago
  48. e92f409 Update WebRTC code version (2024-04-28T04:02:16). by webrtc-version-updater · 11 months ago
  49. c75ee61 Update WebRTC code version (2024-04-27T04:07:22). by webrtc-version-updater · 11 months ago
  50. 5ccd44b Remove EncodedData::reference_buffers. by philipel · 11 months ago
  51. 3703b35 Using Ntp times for the absolute send time. by Jesús de Vicente Peña · 11 months ago
  52. a130e37 Reland "lets try again" by Christoffer Dewerin · 11 months ago
  53. cfddbfe Revert "lets try again" by Christoffer Dewerin · 11 months ago
  54. f03b06e lets try again by Christoffer Dewerin · 11 months ago
  55. 0d9e83c testing by Christoffer Dewerin · 11 months ago
  56. b386d47 Update WebRTC code version (2024-04-26T04:03:31). by webrtc-version-updater · 11 months ago
  57. decc48f Fix 'Screen flickering on ScreenCapturerWinDirectx' by memetao · 1 year, 3 months ago
  58. 3772354 Roll chromium_revision ddd32f326f..669d8ffcd7 (1292311:1292426) by chromium-webrtc-autoroll · 11 months ago
  59. b5f2442 dcsctp: Remove dead code by Victor Boivie · 11 months ago
  60. 2e1a2cd Make stats analysis working with empty layers (bitrate=0) by Sergey Silkin · 11 months ago
  61. d009421 Roll chromium_revision 8b3f58c31e..ddd32f326f (1292052:1292311) by chromium-webrtc-autoroll · 11 months ago
  62. b85b4c0 Reland "New video encoder API." by philipel · 11 months ago
  63. b0e7057 Introduce the TransformerHost interface by Harald Alvestrand · 11 months ago
  64. 28d07dd dcsctp: Compute RTO with higher precision by Victor Boivie · 11 months ago
  65. 1a3120f Move some integration test functions to the .cc file by Harald Alvestrand · 11 months ago
  66. f9a5ed0 Update WebRTC code version (2024-04-25T04:03:46). by webrtc-version-updater · 11 months ago
  67. caa1201 Roll chromium_revision f24efc069c..8b3f58c31e (1291744:1292052) by chromium-webrtc-autoroll · 11 months ago
  68. db50b03 Add perfetto build config by Evan Shrubsole · 11 months ago
  69. 2a66531 Delete deprecated CreateVideoEncoderSoftwareFallbackWrapper by Danil Chapovalov · 11 months ago
  70. c97d434 sdp: cleanup WebRTC-PreventSsrcGroupsWithUnexpectedSize killswitch by Philipp Hancke · 11 months ago
  71. e92b143 Remove VideoCodingModule dependency on the global field trial string by Danil Chapovalov · 11 months ago
  72. af3dfd8 Make WeakPtr slightly cheaper to allocate and use by Tommi · 11 months ago
  73. 56b1799 Roll chromium_revision 1629a193fd..f24efc069c (1291624:1291744) by chromium-webrtc-autoroll · 11 months ago
  74. e4ccad3 Update WebRTC code version (2024-04-24T04:08:12). by webrtc-version-updater · 11 months ago
  75. 4117d19 Roll chromium_revision 5027b7cb30..1629a193fd (1291492:1291624) by chromium-webrtc-autoroll · 11 months ago
  76. 0a8703b Roll chromium_revision 662ec7605b..5027b7cb30 (1291317:1291492) by chromium-webrtc-autoroll · 11 months ago
  77. 4fc2345 Remove IceTransportInternal::SignalGatheringState by Harald Alvestrand · 11 months ago
  78. f1847a1 Roll chromium_revision 84072da101..662ec7605b (1291202:1291317) by chromium-webrtc-autoroll · 11 months ago
  79. fffd489 Add VideoFrameBuffer::storage_presentation. by Markus Handell · 11 months ago
  80. 00a8839 Allow source tracker to be called synchronously on a single thread. by Jakob Ivarsson · 11 months ago
  81. dc3cdf9 Roll chromium_revision b57dda5f8e..84072da101 (1290713:1291202) by chromium-webrtc-autoroll · 11 months ago
  82. 54dec3f Delete deprecated variants for parsing/building AudioLevelExtension by Danil Chapovalov · 11 months ago
  83. bc5c5e9 Migrate webrtc to stop using its own JniZero mirror classes by Mohamed · 11 months ago
  84. 454d651 Fix build errors on GCC w/ libstdc++ 13.2.1 missing cstdint by Vinzenz Feenstra · 12 months ago
  85. 81f09d3 Support all plots in RTC event log analyzer bindings by Björn Terelius · 11 months ago
  86. 3e7d35c Add thread checks to FifoBuffer (test-only class) by Tommi · 11 months ago
  87. 5bfcc87 Add event scope to all TRACE_EVENT_INSTANTs by Evan Shrubsole · 11 months ago
  88. 00566ec Non-inline functions that call CallbackList and are called from Chrome by Harald Alvestrand · 11 months ago
  89. 58cccc6 Cleanup expired experiment WebRTC-SCM-Timestamp by Per K · 11 months ago
  90. cc3ce28 Update WebRTC code version (2024-04-23T04:02:31). by webrtc-version-updater · 11 months ago
  91. d200488 Introduce StreamInterface::FireEvent for firing stream events by Tommi · 11 months ago
  92. 622ca1a stats: fix remote-outbound-rtp id for video by Philipp Hancke · 11 months ago
  93. 15f40ec Roll chromium_revision 7ea464a976..b57dda5f8e (1290570:1290713) by chromium-webrtc-autoroll · 11 months ago
  94. 86298f7 Implementation of RFC 8888 TranportLayerFeedback RTCP packet by Per K · 11 months ago
  95. 6f170a0 Convert P2PtransportChannel.GatheringState to CallbackList by Harald Alvestrand · 11 months ago
  96. 5bd0a32 Roll chromium_revision aaee641ea7..7ea464a976 (1290470:1290570) by chromium-webrtc-autoroll · 11 months ago
  97. f4673f9 Move webrtc::AudioDeviceModule include to api/ folder by Florent Castelli · 11 months ago
  98. f54e013 Remove deprecated ProxyInfo code by Tommi · 11 months ago
  99. cca6cee Remove a couple of deprecated and unused AudioFrameOperations methods by Tommi · 11 months ago
  100. 8d34912 Update WebRTC code version (2024-04-22T04:06:26). by webrtc-version-updater · 11 months ago