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  1. a3796c8 Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap. by Christoffer Rodbro · 2 years, 11 months ago
  2. ce3b3ba Update WebRTC code version (2021-06-17T04:05:50). by webrtc-version-updater · 2 years, 11 months ago
  3. 4b62952 Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293) by chromium-webrtc-autoroll · 2 years, 11 months ago
  4. e0c7365 Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176) by chromium-webrtc-autoroll · 2 years, 11 months ago
  5. a2a073b Reformat pc/g3doc/rtp.md by Artem Titov · 2 years, 11 months ago
  6. 55107c8 Update the sync_group id without recreating audio receive streams. by Tommi · 2 years, 11 months ago
  7. 25029c4 Roll chromium_revision b452ca696d..19c2bebe7d (892948:893060) by chromium-webrtc-autoroll · 2 years, 11 months ago
  8. 355c473 Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. by philipel · 2 years, 11 months ago
  9. 5b9d0c7 AGC1 add clipping predictor evaluator by Alessio Bazzica · 2 years, 11 months ago
  10. 808f494 LOG DTLS (failed) handshake retransmission by Jonas Oreland · 2 years, 11 months ago
  11. d579e6b dcsctp: Do explicit bounds checking in bounded IO by Victor Boivie · 2 years, 11 months ago
  12. 72b7998 Remove the `createDecoder(String)` overload by Xavier Lepaul · 2 years, 11 months ago
  13. 130e031 Roll chromium_revision 570a173256..b452ca696d (892156:892948) by chromium-webrtc-autoroll · 2 years, 11 months ago
  14. 98ff028 AGC analog ClippingPredictor refactoring 2/2 by Alessio Bazzica · 2 years, 11 months ago
  15. 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 2 years, 11 months ago
  16. bc03259 Define generate_location_tags gn arg by Björn Terelius · 2 years, 11 months ago
  17. 6a0a559 Reland "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 2 years, 11 months ago
  18. c03d6e9 Support Java_Buffer_toI420 returning null by Fabian Bergmark · 2 years, 11 months ago
  19. cd430c8 Update WebRTC code version (2021-06-16T04:05:58). by webrtc-version-updater · 2 years, 11 months ago
  20. d6957c2 Revert "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 2 years, 11 months ago
  21. e9ae472 Correctly handle retransmissions/padding in early loss detection. by Erik Språng · 2 years, 11 months ago
  22. e3ceb88 Sanitize hostname literals when mDNS obfuscation is on. by Harald Alvestrand · 2 years, 11 months ago
  23. be53049 Reland "Avoid sending empty receiver reports with RtcpTransceiver" by Danil Chapovalov · 2 years, 11 months ago
  24. 7a2db8a Modify Bundle logic to not add & destroy extra transport at add-track by Harald Alvestrand · 2 years, 11 months ago
  25. e4eb8af libstdc++: fix ostream operator<< usage in JsepTransportCollection by Stephan Hartmann · 2 years, 11 months ago
  26. 07bf5b5 Update WebRTC code version (2021-06-15T04:04:38). by webrtc-version-updater · 2 years, 11 months ago
  27. 3008bcd Don't recreate audio receive streams on header extension update. by Tommi · 2 years, 11 months ago
  28. 6bbe1a4 Roll chromium_revision e9261a56ad..570a173256 (892013:892156) by chromium-webrtc-autoroll · 2 years, 11 months ago
  29. d350006 Add rtp_config() accessor to ReceiveStream. by Tommi · 2 years, 11 months ago
  30. 48420fa Revert "Avoid sending empty receiver reports with RtcpTransceiver" by Björn Terelius · 2 years, 11 months ago
  31. 1c1f540 Factor out common receive stream methods to a common interface. by Tommi · 2 years, 11 months ago
  32. e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 2 years, 11 months ago
  33. e5f1a39 Avoid sending empty receiver reports with RtcpTransceiver by Danil Chapovalov · 2 years, 11 months ago
  34. 8b69290 Fix VideoStreamEncoder QP tests to not use SetHasInternalSource by Niels Möller · 2 years, 11 months ago
  35. b237a87 AGC analog ClippingPredictor refactoring 1/2 by Alessio Bazzica · 2 years, 11 months ago
  36. 1ff491b Roll chromium_revision 8907aace7e..e9261a56ad (891631:892013) by chromium-webrtc-autoroll · 2 years, 11 months ago
  37. 74cc9ea Don't register invalid encode complete callbacks. by Peter Hanspers · 2 years, 11 months ago
  38. 1081487 Avoid video stream allocation on configuration change after timeout. by Jakob Ivarsson · 2 years, 11 months ago
  39. 13dac0c Update WebRTC code version (2021-06-13T04:02:18). by webrtc-version-updater · 2 years, 11 months ago
  40. 0f9a8e33 Make stopping of the RepeatingTask safer by Danil Chapovalov · 2 years, 11 months ago
  41. 4bb81ac Make JsepTransportCollection self-managing for transports by Harald Alvestrand · 2 years, 11 months ago
  42. 63c96ce Roll chromium_revision d2f297f391..8907aace7e (890623:891631) by chromium-webrtc-autoroll · 2 years, 11 months ago
  43. 8d3396d In vp9 encoder fuzzer reduce information stored for older frames by Danil Chapovalov · 2 years, 11 months ago
  44. a63d152 AEC3: Unbounded echo spectrum for dominant nearend detection. by Gustaf Ullberg · 2 years, 11 months ago
  45. 1b4807f count webrtc pranswer usage by Philipp Hancke · 2 years, 11 months ago
  46. b22abbc Add kron as owner of api/uma_metrics.h by Johannes Kron · 2 years, 11 months ago
  47. ef4edaf Remove DEPS that was removed from chromium by Andrey Logvin · 2 years, 11 months ago
  48. ec6b655 Break out pc/session_description build target (part 2) by Harald Alvestrand · 2 years, 11 months ago
  49. f5f7e8e Ensure that fps adaptation count can go back to zero when framerate is unrestricted. by Åsa Persson · 2 years, 11 months ago
  50. c63ae48 Prepare for breakout of session_description.{h,cc} by Harald Alvestrand · 2 years, 11 months ago
  51. 62ec0f6 Add small cooldown to unsignalled ssrc stream creation. by Henrik Boström · 2 years, 11 months ago
  52. ba7da8b Relax expectation in OveruseFrameDetectorTest2.ConvergesSlowly by Niels Möller · 2 years, 11 months ago
  53. 9dea393 Move MID/JsepTransport mappings into a new manager object. by Harald Alvestrand · 2 years, 11 months ago
  54. 64e3a36 Update WebRTC code version (2021-06-10T04:05:52). by webrtc-version-updater · 2 years, 11 months ago
  55. 5e65dd5 Add MB configs for M1 bots by Christoffer Jansson · 2 years, 11 months ago
  56. 3cc68ec Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz. by Tommi · 2 years, 11 months ago
  57. e2e0464 Remove a couple of locks from ChannelReceive and add thread checks. by Tommi · 2 years, 11 months ago
  58. b56a63e dcsctp: Prevent overflow of missing parameters by Victor Boivie · 2 years, 11 months ago
  59. 6eda26c Reland "Remove AudioReceiveStream::Reconfigure() method." by Tommi · 2 years, 11 months ago
  60. c0a9586 Break out pc/session_description in its own build target (part 1) by Harald Alvestrand · 2 years, 11 months ago
  61. 8a18e5b Revert "Remove AudioReceiveStream::Reconfigure() method." by Andrey Logvin · 2 years, 11 months ago
  62. e2561e1 Remove AudioReceiveStream::Reconfigure() method. by Tommi · 2 years, 11 months ago
  63. 4ea80f3 Disable PT based demuxing if MID header extension is present. by Henrik Boström · 2 years, 11 months ago
  64. 33e75bb Roll chromium_revision 415567486a..d2f297f391 (890510:890623) by chromium-webrtc-autoroll · 2 years, 11 months ago
  65. 80d6669 Roll chromium_revision 94153b0fd3..415567486a (890203:890510) by chromium-webrtc-autoroll · 2 years, 11 months ago
  66. 58126f9 Update the only 3 remaining kFilterBilinear to kFilterBox. by Henrik Boström · 2 years, 11 months ago
  67. d9a135b Roll chromium_revision a72439959b..94153b0fd3 (889958:890203) by chromium-webrtc-autoroll · 2 years, 11 months ago
  68. 2aa24f1 Move group-modifying functions into BundleManager by Harald Alvestrand · 2 years, 11 months ago
  69. 9f9bf38 Start refactoring bundle behavior into BundleManager by Harald Alvestrand · 2 years, 11 months ago
  70. 2e15468 Avoid generating a random id for candidate stats. by Tommi · 2 years, 11 months ago
  71. 0fb9497 Update WebRTC code version (2021-06-08T04:04:02). by webrtc-version-updater · 2 years, 11 months ago
  72. c6f2fef Roll chromium_revision 98040cdbe1..a72439959b (889744:889958) by chromium-webrtc-autoroll · 2 years, 11 months ago
  73. da8a45f AllocationSequence: migrate from rtc::Message to TaskQueue. by Markus Handell · 2 years, 11 months ago
  74. dedcdfe AllocationSequence: switch signal to callback. by Markus Handell · 2 years, 11 months ago
  75. fd89fc7 BasicPortAllocatorSession: migrate to TaskQueue. by Markus Handell · 2 years, 11 months ago
  76. 637a9ee Roll chromium_revision 8d359ae542..98040cdbe1 (889625:889744) by chromium-webrtc-autoroll · 2 years, 11 months ago
  77. 518669d Add more trace events to interesting places. by Markus Handell · 2 years, 11 months ago
  78. a94a4cc Roll chromium_revision 2951ce9ba1..8d359ae542 (889518:889625) by chromium-webrtc-autoroll · 2 years, 11 months ago
  79. 5032f54 Update WebRTC code version (2021-06-06T04:02:33). by webrtc-version-updater · 2 years, 11 months ago
  80. b60ebfe Roll chromium_revision f54fe52cfb..2951ce9ba1 (889417:889518) by chromium-webrtc-autoroll · 2 years, 11 months ago
  81. b6d50e3 Handle encoder_ == nullptr in VideoStreamEncoder::EncodeVideoFrame. by Tommi · 2 years, 11 months ago
  82. f277300 Roll chromium_revision 44fa1f9723..f54fe52cfb (889277:889417) by chromium-webrtc-autoroll · 2 years, 11 months ago
  83. 1a778a2 Avoid using legacy rtp header parser in the rtp_to_text tool by Danil Chapovalov · 3 years ago
  84. 1b63db9 Move AV1X-AV1 mapping to VideoCodecTypeMime by Sergey Silkin · 2 years, 11 months ago
  85. e34380a Roll chromium_revision b54a8c30e7..44fa1f9723 (889150:889277) by chromium-webrtc-autoroll · 2 years, 11 months ago
  86. a334dc6 Make VideoSendStream::UpdateActiveSimulcastLayers not block. by Tommi · 2 years, 11 months ago
  87. d25af8ce doc: document rtp payload type mapping behaviour by Philipp Hancke · 2 years, 11 months ago
  88. bc8e175 Roll chromium_revision a11573a242..b54a8c30e7 (888935:889150) by chromium-webrtc-autoroll · 2 years, 11 months ago
  89. ffbfba9 Added `PeerConnectionObserverJni::OnRemoveTrack()` by Jesús Leganés-Combarro 'piranna · 3 years ago
  90. 1050fbc Remove synchronization from VideoSendStream construction. by Tommi · 3 years ago
  91. c2b4d9b Roll chromium_revision fb5254ac9f..a11573a242 (888818:888935) by chromium-webrtc-autoroll · 3 years ago
  92. 52c7fd6 Modernize style in RemoteBitrateEstimatorAbsSendTime implementation by Danil Chapovalov · 3 years ago
  93. 43eb4f5 Roll chromium_revision fee5f397ef..fb5254ac9f (888712:888818) by chromium-webrtc-autoroll · 3 years ago
  94. 0960143 Add a function to check if the packet in a `PacketResult` has been received. by Fanny Linderborg · 3 years ago
  95. 47f5f8c Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket by Danil Chapovalov · 3 years ago
  96. 943e2e6 Revert "Fix incorrect SSRC in RtpPacketSendInfo for RTX packets." by Andrey Logvin · 3 years ago
  97. fa3ce63 Simplify VideoSendStreamImpl constructor. by Tommi · 3 years ago
  98. 84f4ca6 Remove workaround for broken nearby build in Chrome by Evan Shrubsole · 3 years ago
  99. e902f28 Make VideoSendStreamImpl::configured_pacing_factor_ const by Tommi · 3 years ago
  100. 28e9653 Remove dependency on RtpVideoSenderInterface from EncoderRtcpFeedback. by Tommi · 3 years ago