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chromium/external/webrtc/HEAD
  1. 13dac0c Update WebRTC code version (2021-06-13T04:02:18). by webrtc-version-updater · 4 years, 8 months ago
  2. 0f9a8e3 Make stopping of the RepeatingTask safer by Danil Chapovalov · 4 years, 8 months ago
  3. 4bb81ac Make JsepTransportCollection self-managing for transports by Harald Alvestrand · 4 years, 8 months ago
  4. 63c96ce Roll chromium_revision d2f297f391..8907aace7e (890623:891631) by chromium-webrtc-autoroll · 4 years, 8 months ago
  5. 8d3396d In vp9 encoder fuzzer reduce information stored for older frames by Danil Chapovalov · 4 years, 8 months ago
  6. a63d152 AEC3: Unbounded echo spectrum for dominant nearend detection. by Gustaf Ullberg · 4 years, 8 months ago
  7. 1b4807f count webrtc pranswer usage by Philipp Hancke · 4 years, 8 months ago
  8. b22abbc Add kron as owner of api/uma_metrics.h by Johannes Kron · 4 years, 8 months ago
  9. ef4edaf Remove DEPS that was removed from chromium by Andrey Logvin · 4 years, 8 months ago
  10. ec6b655 Break out pc/session_description build target (part 2) by Harald Alvestrand · 4 years, 8 months ago
  11. f5f7e8e Ensure that fps adaptation count can go back to zero when framerate is unrestricted. by Åsa Persson · 4 years, 8 months ago
  12. c63ae48 Prepare for breakout of session_description.{h,cc} by Harald Alvestrand · 4 years, 8 months ago
  13. 62ec0f6 Add small cooldown to unsignalled ssrc stream creation. by Henrik Boström · 4 years, 8 months ago
  14. ba7da8b Relax expectation in OveruseFrameDetectorTest2.ConvergesSlowly by Niels Möller · 4 years, 8 months ago
  15. 9dea393 Move MID/JsepTransport mappings into a new manager object. by Harald Alvestrand · 4 years, 8 months ago
  16. 64e3a36 Update WebRTC code version (2021-06-10T04:05:52). by webrtc-version-updater · 4 years, 8 months ago
  17. 5e65dd5 Add MB configs for M1 bots by Christoffer Jansson · 4 years, 8 months ago
  18. 3cc68ec Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz. by Tommi · 4 years, 8 months ago
  19. e2e0464 Remove a couple of locks from ChannelReceive and add thread checks. by Tommi · 4 years, 8 months ago
  20. b56a63e dcsctp: Prevent overflow of missing parameters by Victor Boivie · 4 years, 8 months ago
  21. 6eda26c Reland "Remove AudioReceiveStream::Reconfigure() method." by Tommi · 4 years, 8 months ago
  22. c0a9586 Break out pc/session_description in its own build target (part 1) by Harald Alvestrand · 4 years, 8 months ago
  23. 8a18e5b Revert "Remove AudioReceiveStream::Reconfigure() method." by Andrey Logvin · 4 years, 8 months ago
  24. e2561e1 Remove AudioReceiveStream::Reconfigure() method. by Tommi · 4 years, 8 months ago
  25. 4ea80f3 Disable PT based demuxing if MID header extension is present. by Henrik Boström · 4 years, 8 months ago
  26. 33e75bb Roll chromium_revision 415567486a..d2f297f391 (890510:890623) by chromium-webrtc-autoroll · 4 years, 8 months ago
  27. 80d6669 Roll chromium_revision 94153b0fd3..415567486a (890203:890510) by chromium-webrtc-autoroll · 4 years, 8 months ago
  28. 58126f9 Update the only 3 remaining kFilterBilinear to kFilterBox. by Henrik Boström · 4 years, 8 months ago
  29. d9a135b Roll chromium_revision a72439959b..94153b0fd3 (889958:890203) by chromium-webrtc-autoroll · 4 years, 8 months ago
  30. 2aa24f1 Move group-modifying functions into BundleManager by Harald Alvestrand · 4 years, 8 months ago
  31. 9f9bf38 Start refactoring bundle behavior into BundleManager by Harald Alvestrand · 4 years, 8 months ago
  32. 2e15468 Avoid generating a random id for candidate stats. by Tommi · 4 years, 8 months ago
  33. 0fb9497 Update WebRTC code version (2021-06-08T04:04:02). by webrtc-version-updater · 4 years, 8 months ago
  34. c6f2fef Roll chromium_revision 98040cdbe1..a72439959b (889744:889958) by chromium-webrtc-autoroll · 4 years, 8 months ago
  35. da8a45f AllocationSequence: migrate from rtc::Message to TaskQueue. by Markus Handell · 4 years, 8 months ago
  36. dedcdfe AllocationSequence: switch signal to callback. by Markus Handell · 4 years, 8 months ago
  37. fd89fc7 BasicPortAllocatorSession: migrate to TaskQueue. by Markus Handell · 4 years, 8 months ago
  38. 637a9ee Roll chromium_revision 8d359ae542..98040cdbe1 (889625:889744) by chromium-webrtc-autoroll · 4 years, 8 months ago
  39. 518669d Add more trace events to interesting places. by Markus Handell · 4 years, 8 months ago
  40. a94a4cc Roll chromium_revision 2951ce9ba1..8d359ae542 (889518:889625) by chromium-webrtc-autoroll · 4 years, 8 months ago
  41. 5032f54 Update WebRTC code version (2021-06-06T04:02:33). by webrtc-version-updater · 4 years, 8 months ago
  42. b60ebfe Roll chromium_revision f54fe52cfb..2951ce9ba1 (889417:889518) by chromium-webrtc-autoroll · 4 years, 9 months ago
  43. b6d50e3 Handle encoder_ == nullptr in VideoStreamEncoder::EncodeVideoFrame. by Tommi · 4 years, 9 months ago
  44. f277300 Roll chromium_revision 44fa1f9723..f54fe52cfb (889277:889417) by chromium-webrtc-autoroll · 4 years, 9 months ago
  45. 1a778a2 Avoid using legacy rtp header parser in the rtp_to_text tool by Danil Chapovalov · 4 years, 9 months ago
  46. 1b63db9 Move AV1X-AV1 mapping to VideoCodecTypeMime by Sergey Silkin · 4 years, 9 months ago
  47. e34380a Roll chromium_revision b54a8c30e7..44fa1f9723 (889150:889277) by chromium-webrtc-autoroll · 4 years, 9 months ago
  48. a334dc6 Make VideoSendStream::UpdateActiveSimulcastLayers not block. by Tommi · 4 years, 9 months ago
  49. d25af8ce doc: document rtp payload type mapping behaviour by Philipp Hancke · 4 years, 9 months ago
  50. bc8e175 Roll chromium_revision a11573a242..b54a8c30e7 (888935:889150) by chromium-webrtc-autoroll · 4 years, 9 months ago
  51. ffbfba9 Added `PeerConnectionObserverJni::OnRemoveTrack()` by Jesús Leganés-Combarro 'piranna · 4 years, 9 months ago
  52. 1050fbc Remove synchronization from VideoSendStream construction. by Tommi · 4 years, 9 months ago
  53. c2b4d9b Roll chromium_revision fb5254ac9f..a11573a242 (888818:888935) by chromium-webrtc-autoroll · 4 years, 9 months ago
  54. 52c7fd6 Modernize style in RemoteBitrateEstimatorAbsSendTime implementation by Danil Chapovalov · 4 years, 9 months ago
  55. 43eb4f5 Roll chromium_revision fee5f397ef..fb5254ac9f (888712:888818) by chromium-webrtc-autoroll · 4 years, 9 months ago
  56. 0960143 Add a function to check if the packet in a `PacketResult` has been received. by Fanny Linderborg · 4 years, 9 months ago
  57. 47f5f8c Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket by Danil Chapovalov · 4 years, 9 months ago
  58. 943e2e6 Revert "Fix incorrect SSRC in RtpPacketSendInfo for RTX packets." by Andrey Logvin · 4 years, 9 months ago
  59. fa3ce63 Simplify VideoSendStreamImpl constructor. by Tommi · 4 years, 9 months ago
  60. 84f4ca6 Remove workaround for broken nearby build in Chrome by Evan Shrubsole · 4 years, 9 months ago
  61. e902f28 Make VideoSendStreamImpl::configured_pacing_factor_ const by Tommi · 4 years, 9 months ago
  62. 28e9653 Remove dependency on RtpVideoSenderInterface from EncoderRtcpFeedback. by Tommi · 4 years, 9 months ago
  63. 1d2b22e Use pixels from single active stream if set for balanced degradation settings. by Åsa Persson · 4 years, 9 months ago
  64. 2a25a96 Disable flacky tests on mac bots by Ilya Nikolaevskiy · 4 years, 9 months ago
  65. f3ff3c5 Reinstate killswitch for WebRTC-Bwe-ReceiverLimitCapsOnly. by Christoffer Rodbro · 4 years, 9 months ago
  66. ab229b0 Add documentation for RTC event log by Björn Terelius · 4 years, 9 months ago
  67. 31b5649 Update comment for RtpVideoStreamReceiver2::RequestPacketRetransmit. by Tommi · 4 years, 9 months ago
  68. e93dc74 Roll chromium_revision c165693ba5..fee5f397ef (888523:888712) by chromium-webrtc-autoroll · 4 years, 9 months ago
  69. a004715 Integrate ClippingPredictor into AudioProcessingImpl and AgcManagerDirect by Hanna Silen · 4 years, 9 months ago
  70. 4b3a061 Add ClippingPredictor implementation by Hanna Silen · 4 years, 9 months ago
  71. 565ad61 Roll chromium_revision 936a99501f..c165693ba5 (888404:888523) by chromium-webrtc-autoroll · 4 years, 9 months ago
  72. a43953a Add ClippingPredictor config in AudioProcessing config by Hanna Silen · 4 years, 9 months ago
  73. cbdbb8c Add ability to adjust the suppressor smoothing in AEC3 by Per Åhgren · 4 years, 9 months ago
  74. bd933ee SdpOfferAnswerHandler: Significantly reduce audio impairment. by Markus Handell · 4 years, 9 months ago
  75. 7444b19 Add integration test for active stream toggling. by Erik Språng · 4 years, 9 months ago
  76. 4410789 Roll chromium_revision e53f664c6c..936a99501f (888291:888404) by chromium-webrtc-autoroll · 4 years, 9 months ago
  77. fccb052 Add event traces to interesting places in WebRTC. by Markus Handell · 4 years, 9 months ago
  78. 486b040 Make VP8 DefaultTemporalLayers always report TL count even with no rate. by Erik Språng · 4 years, 9 months ago
  79. 1c7ff0d dcsctp: Stay in stream if not producing fragment by Victor Boivie · 4 years, 9 months ago
  80. 5981bf2 Add resolution alignment properties to RTCVideoEncoder protocol. by Peter Hanspers · 4 years, 9 months ago
  81. aaa835c Update WebRTC code version (2021-06-02T04:02:07). by webrtc-version-updater · 4 years, 9 months ago
  82. f32b400 Roll chromium_revision 45bbaf2c3c..e53f664c6c (888151:888291) by chromium-webrtc-autoroll · 4 years, 9 months ago
  83. 62678f5 Roll chromium_revision 10da87c3f6..45bbaf2c3c (888035:888151) by chromium-webrtc-autoroll · 4 years, 9 months ago
  84. bf952fa Roll chromium_revision ed24ed8d5d..10da87c3f6 (887902:888035) by chromium-webrtc-autoroll · 4 years, 9 months ago
  85. 78c7347 Add DesktopCaptureOption enumerate_current_process_windows to avoid hang by Austin Orion · 4 years, 9 months ago
  86. 803fdc4 dcsctp: Stay within stream while producing from it by Victor Boivie · 4 years, 9 months ago
  87. f865444 Make AV1 respect spatial layer active flag. by Erik Språng · 4 years, 9 months ago
  88. d23628d Remove RecordingState::keyframe_needed. by Tommi · 4 years, 9 months ago
  89. d994304 Call: introduce SendStats. by Markus Handell · 4 years, 9 months ago
  90. e9fa954 Roll chromium_revision 03cca1960d..ed24ed8d5d (887795:887902) by chromium-webrtc-autoroll · 4 years, 9 months ago
  91. 7594145 Fix incorrect fps_allocation printed by EncoderInfo::ToString() by Erik Språng · 4 years, 9 months ago
  92. 5cb983b Add basic synchronization function info to g3doc by Harald Alvestrand · 4 years, 9 months ago
  93. 3907e7b AudioSendStream: s/worker_queue_/rtp_transport_queue_/g by Markus Handell · 4 years, 9 months ago
  94. 58b8d29 fall back to payload types from lower range after exhausting [96,127] by Philipp Hancke · 4 years, 9 months ago
  95. 504fc19 Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. by Vojin Ilic · 4 years, 9 months ago
  96. 40f1a06 Update WebRTC code version (2021-06-01T04:04:24). by webrtc-version-updater · 4 years, 9 months ago
  97. 7d23535 Populate qualityLimitationDurations stats for outbound RTP streams by Byoungchan Lee · 4 years, 9 months ago
  98. 7b4fd5c dcsctp: Determine chunks to be retransmitted fast by Victor Boivie · 4 years, 9 months ago
  99. 82aa094 Fix incorrect SSRC in RtpPacketSendInfo for RTX packets. by memetao · 4 years, 9 months ago
  100. c48a49c dcsctp: Find out quickly if to send FORWARD-TSN by Victor Boivie · 4 years, 9 months ago