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  1. 1d2b22e Use pixels from single active stream if set for balanced degradation settings. by Åsa Persson · 3 years ago
  2. 2a25a96 Disable flacky tests on mac bots by Ilya Nikolaevskiy · 3 years ago
  3. f3ff3c5 Reinstate killswitch for WebRTC-Bwe-ReceiverLimitCapsOnly. by Christoffer Rodbro · 3 years ago
  4. ab229b0 Add documentation for RTC event log by Björn Terelius · 3 years ago
  5. 31b5649 Update comment for RtpVideoStreamReceiver2::RequestPacketRetransmit. by Tommi · 3 years ago
  6. e93dc74 Roll chromium_revision c165693ba5..fee5f397ef (888523:888712) by chromium-webrtc-autoroll · 3 years ago
  7. a004715 Integrate ClippingPredictor into AudioProcessingImpl and AgcManagerDirect by Hanna Silen · 3 years ago
  8. 4b3a061 Add ClippingPredictor implementation by Hanna Silen · 3 years ago
  9. 565ad61 Roll chromium_revision 936a99501f..c165693ba5 (888404:888523) by chromium-webrtc-autoroll · 3 years ago
  10. a43953a Add ClippingPredictor config in AudioProcessing config by Hanna Silen · 3 years ago
  11. cbdbb8c Add ability to adjust the suppressor smoothing in AEC3 by Per Åhgren · 3 years ago
  12. bd933ee SdpOfferAnswerHandler: Significantly reduce audio impairment. by Markus Handell · 3 years ago
  13. 7444b19 Add integration test for active stream toggling. by Erik Språng · 3 years ago
  14. 4410789 Roll chromium_revision e53f664c6c..936a99501f (888291:888404) by chromium-webrtc-autoroll · 3 years ago
  15. fccb052 Add event traces to interesting places in WebRTC. by Markus Handell · 3 years ago
  16. 486b040 Make VP8 DefaultTemporalLayers always report TL count even with no rate. by Erik Språng · 3 years ago
  17. 1c7ff0d dcsctp: Stay in stream if not producing fragment by Victor Boivie · 3 years ago
  18. 5981bf2 Add resolution alignment properties to RTCVideoEncoder protocol. by Peter Hanspers · 3 years ago
  19. aaa835c Update WebRTC code version (2021-06-02T04:02:07). by webrtc-version-updater · 3 years ago
  20. f32b400 Roll chromium_revision 45bbaf2c3c..e53f664c6c (888151:888291) by chromium-webrtc-autoroll · 3 years ago
  21. 62678f5 Roll chromium_revision 10da87c3f6..45bbaf2c3c (888035:888151) by chromium-webrtc-autoroll · 3 years ago
  22. bf952fa Roll chromium_revision ed24ed8d5d..10da87c3f6 (887902:888035) by chromium-webrtc-autoroll · 3 years ago
  23. 78c7347 Add DesktopCaptureOption enumerate_current_process_windows to avoid hang by Austin Orion · 3 years ago
  24. 803fdc4 dcsctp: Stay within stream while producing from it by Victor Boivie · 3 years ago
  25. f8654448 Make AV1 respect spatial layer active flag. by Erik Språng · 3 years ago
  26. d23628d Remove RecordingState::keyframe_needed. by Tommi · 3 years ago
  27. d994304 Call: introduce SendStats. by Markus Handell · 3 years ago
  28. e9fa954 Roll chromium_revision 03cca1960d..ed24ed8d5d (887795:887902) by chromium-webrtc-autoroll · 3 years ago
  29. 7594145 Fix incorrect fps_allocation printed by EncoderInfo::ToString() by Erik Språng · 3 years ago
  30. 5cb983b Add basic synchronization function info to g3doc by Harald Alvestrand · 3 years ago
  31. 3907e7b AudioSendStream: s/worker_queue_/rtp_transport_queue_/g by Markus Handell · 3 years ago
  32. 58b8d29 fall back to payload types from lower range after exhausting [96,127] by Philipp Hancke · 3 years ago
  33. 504fc19 Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. by Vojin Ilic · 3 years ago
  34. 40f1a06 Update WebRTC code version (2021-06-01T04:04:24). by webrtc-version-updater · 3 years ago
  35. 7d23535 Populate qualityLimitationDurations stats for outbound RTP streams by Byoungchan Lee · 3 years ago
  36. 7b4fd5c dcsctp: Determine chunks to be retransmitted fast by Victor Boivie · 3 years ago
  37. 82aa094 Fix incorrect SSRC in RtpPacketSendInfo for RTX packets. by memetao · 3 years ago
  38. c48a49c dcsctp: Find out quickly if to send FORWARD-TSN by Victor Boivie · 3 years ago
  39. cc84c98 Fix typo in a URL in the comment by Byoungchan Lee · 3 years ago
  40. 376cf07 Add packet_sequence_checker_ to RtpVideoStreamReceiver2. by Tommi · 3 years ago
  41. b6b7d80 Roll chromium_revision fbd3997e87..03cca1960d (887673:887795) by chromium-webrtc-autoroll · 3 years ago
  42. 90738dd Split VideoReceiveStream2 init into worker / network steps. by Tommi · 3 years ago
  43. 27d2be3 dcsctp: Optimize SACK generation by Victor Boivie · 3 years ago
  44. 0377bab Split FlexfecReceiveStreamImpl init into worker / network steps. by Tommi · 3 years ago
  45. 261eec5 dcsctp: Allow more outstanding fragments by Victor Boivie · 3 years ago
  46. 8267724 dcsctp: Announce send buffer watermark as a_rwnd by Victor Boivie · 3 years ago
  47. ea72ee6 Add ClippingPredictorLevelBuffer circular buffer. by Hanna Silen · 3 years ago
  48. 4f26a3c red: assign payload type 63 to audio/RED for opus by Philipp Hancke · 3 years ago
  49. 5d4c3c5 dcsctp: Add more unit tests for DataTracker by Victor Boivie · 3 years ago
  50. 5429d71 dcsctp: Allow heartbeats to be disabled by Victor Boivie · 3 years ago
  51. 02df2eb Split AudioStream initialization into worker / network steps. by Tommi · 3 years ago
  52. 6ad542c Remove temporary using webrtc::OnCompleteFrameCallback statement. by philipel · 3 years ago
  53. 948e40c Add thread guards and constness to Call members. by Tommi · 3 years ago
  54. cae1f1d Move PostTask for DeliverRtcp from PeerConnection to Call. by Tommi · 3 years ago
  55. acd16af AudioReceiveStream: Clean up ConfigureStream. by Markus Handell · 3 years ago
  56. c81afe3 Call: prepare receive stats for thread switch. by Markus Handell · 3 years ago
  57. 22fead3 Roll chromium_revision 6f6904aacc..fbd3997e87 (887571:887673) by chromium-webrtc-autoroll · 3 years ago
  58. 5be2aa1 Make generate_license.py compatible with Python 3. by Byoungchan Lee · 3 years ago
  59. d3166af Update WebRTC code version (2021-05-31T04:03:11). by webrtc-version-updater · 3 years ago
  60. d280eaf Update WebRTC code version (2021-05-30T04:02:15). by webrtc-version-updater · 3 years ago
  61. 6c94d58 Roll chromium_revision bbca8ebcc5..6f6904aacc (887470:887571) by chromium-webrtc-autoroll · 3 years ago
  62. cbb4421 Remove DeliverPacketAsync. by Tommi · 3 years ago
  63. 7857251 Update WebRTC code version (2021-05-29T04:03:30). by webrtc-version-updater · 3 years ago
  64. d5b0199 Roll chromium_revision 4a5a62a362..bbca8ebcc5 (887362:887470) by chromium-webrtc-autoroll · 3 years ago
  65. 3d46d0b Proxy: solve event tracing with compile time strings. by Markus Handell · 3 years ago
  66. d325f32 Update WebRTC code version (2021-05-28T04:03:27). by webrtc-version-updater · 3 years ago
  67. 319bac6 Roll chromium_revision f713d4fb04..4a5a62a362 (887229:887362) by chromium-webrtc-autoroll · 3 years ago
  68. d595f6c Roll chromium_revision a0132a2044..f713d4fb04 (887053:887229) by chromium-webrtc-autoroll · 3 years ago
  69. c39080c Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions by Artem Titov · 3 years ago
  70. 236ac50 dcsctp: Add public API for BufferedAmountLow by Victor Boivie · 3 years ago
  71. bd9031b dcsctp: Add OnTotalBufferedAmountLow in Send Queue by Victor Boivie · 3 years ago
  72. 791adaf dcsctp: Add OnBufferedAmountLow in Send Queue by Victor Boivie · 3 years ago
  73. a1b8201 Move proxies into pc/. by Markus Handell · 3 years ago
  74. 7d2e669 dcsctp: Allocate TSN for end of abandoned message by Victor Boivie · 3 years ago
  75. 9700d88 dcsctp: Avoid recalculation of outstanding bytes by Victor Boivie · 3 years ago
  76. 36ad606 Update WebRTC code version (2021-05-27T04:02:45). by webrtc-version-updater · 3 years ago
  77. e52cfab PipeWire capturer: request mouse cursor to be part of the stream by Jan Grulich · 3 years, 1 month ago
  78. 2182096 RtpFrameReferenceFinder return frames directly instead of via callback. by philipel · 3 years ago
  79. 7f11067 Clean up RtpSenderTest and remove RtpSenderEgress dependencies. by Erik Språng · 3 years ago
  80. b4f3204 Remove Win UWP mb config by Christoffer Jansson · 3 years ago
  81. 940108b Apply autoformat to the docs by Artem Titov · 3 years ago
  82. 8f8bf25 Remove usage of InjectPacket and transport_ in rtp_sender_unittest by Erik Språng · 3 years ago
  83. b412efd payload type mapping: restrict lower range to <= 63 by Philipp Hancke · 3 years ago
  84. a9af50f Introduce CreateDataChannelOrError by Harald Alvestrand · 3 years ago
  85. 0d0ed76 Fix RTP header extension encryption by Lennart Grahl · 3 years ago
  86. 4a54be7 doc: update dtls_transport.md to use new link style by Philipp Hancke · 3 years ago
  87. fec79b7 add srtp docs by Philipp Hancke · 3 years ago
  88. 770acab Refactor mid/rid rtp tests to avoid using egress/transport logic. by Erik Språng · 3 years ago
  89. a39d966 Remove unused property isLocked from RTCAudioSession by Byoungchan Lee · 3 years, 1 month ago
  90. 8d9d575 PipeWire capturer: fix stream width in PW 0.2 code by Jan Grulich · 3 years, 1 month ago
  91. 27df007 Update WebRTC code version (2021-05-26T04:05:14). by webrtc-version-updater · 3 years ago
  92. 048bf18 Roll chromium_revision 8a4c5eb899..19159a8788 (886374:886529) by chromium-webrtc-autoroll · 3 years ago
  93. 0a52ede Support for map of string keys to uint64_t / double values in RTCStats by Byoungchan Lee · 3 years ago
  94. cbeff55 Roll chromium_revision 1b27d646a6..8a4c5eb899 (886225:886374) by chromium-webrtc-autoroll · 3 years ago
  95. 1573716 Enforce thread invoke policy for invokes to itself by Artem Titov · 3 years ago
  96. 4fbc3fc Move SendPacketUpdates* tests to rtp_sender_egress_unittest. by Erik Språng · 3 years ago
  97. fade919 Partial revert: "Use unordered map in RtpDemuxer" by Victor Boivie · 3 years ago
  98. 238da9a Remove obsolete SendPacketMatches* tests from rtp_sender_egress_unittest. by Erik Språng · 3 years ago
  99. 552169c Refactor RtpPacketCounter tests and move to rtp_sender_egress_unittest. by Erik Språng · 3 years ago
  100. af0dff0 dcsctp: start SCTP_DUMP on a new line by Philipp Hancke · 3 years ago