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webrtc / src.git / 0fe60bd41a9afff43a17a4e38194c7cbbc972421
  1. 0fe60bd Add RecursiveCriticalSection to the don't-use list of primitives by Harald Alvestrand · 3 years, 10 months ago
  2. c413c55 Replace use of RecursiveCriticalSection in VirtualSocketServer by Niels Möller · 3 years, 10 months ago
  3. fe6580f Revert "Fix echo return loss stats and add to RTCAudioSourceStats." by Evan Shrubsole · 3 years, 10 months ago
  4. 9e2b315 Minor code cleanup of WebRtcVideoReceiveStream. by Tommi · 3 years, 10 months ago
  5. 885d538 ModuleRtpRtcpImpl2: remove RTCP send polling. by Markus Handell · 3 years, 10 months ago
  6. 2086209 Update WebRTC code version (2021-06-22T04:05:30). by webrtc-version-updater · 3 years, 10 months ago
  7. 049ed44 ModuleRtpRtcpImpl2: update test code. by Markus Handell · 3 years, 10 months ago
  8. fb7fd24 Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering. by Jake Bromberg · 3 years, 10 months ago
  9. c6b9ac7 RTCPSender: migrate to Timestamp. by Markus Handell · 3 years, 10 months ago
  10. e2ab77b Reland "Port: migrate to TaskQueue." by Markus Handell · 3 years, 10 months ago
  11. a27cfbf Fix echo return loss stats and add to RTCAudioSourceStats. by Taylor Brandstetter · 3 years, 10 months ago
  12. 2e3edc1 RTCPSender: migrate to own configuration struct. by Markus Handell · 3 years, 10 months ago
  13. f906ec4 Handle null return from ToI420 in encoders by Evan Shrubsole · 3 years, 10 months ago
  14. 76a35d9 Delete legacy RtpHeaderParser wrapper by Danil Chapovalov · 3 years, 10 months ago
  15. 257f81b Update VirtualSocketServer locking to match documentation. by Niels Möller · 3 years, 10 months ago
  16. a4aabb9 Revert "Port: migrate to TaskQueue." by Markus Handell · 3 years, 10 months ago
  17. 0654016 Port: migrate to TaskQueue. by Markus Handell · 3 years, 10 months ago
  18. 3f6efdc Update WebRTC code version (2021-06-21T04:05:45). by webrtc-version-updater · 3 years, 10 months ago
  19. ae278d4 openssl_adapter: document SSL_CTX_set_verify_depth behaviour by Philipp Hancke · 3 years, 10 months ago
  20. fbe9958 Update WebRTC code version (2021-06-20T04:03:02). by webrtc-version-updater · 3 years, 10 months ago
  21. 4cacaf7 Update WebRTC code version (2021-06-19T04:03:03). by webrtc-version-updater · 3 years, 10 months ago
  22. 7c719b0 Fixes off-by-one error in video capture module by Johannes Kron · 3 years, 10 months ago
  23. bad0ab0 Delete unused class MockDelayable by Niels Möller · 3 years, 10 months ago
  24. c6d76489e Add jakobi to modules/audio_coding OWNERS by Ivo Creusen · 3 years, 10 months ago
  25. 6a11c84 dcsctp: Add DcSctpSocketFactory by Florent Castelli · 3 years, 10 months ago
  26. c20f156 dcsctp: Don't sent more packets before COOKIE ACK by Victor Boivie · 3 years, 10 months ago
  27. 95c3041 Update WebRTC code version (2021-06-18T04:03:27). by webrtc-version-updater · 3 years, 10 months ago
  28. 42dacda AGC analog clipping predictor: integrate evaluator by Alessio Bazzica · 3 years, 10 months ago
  29. 7d54182 Avoid assembling complicated but unused video rtp header extensions by Danil Chapovalov · 3 years, 10 months ago
  30. afb28116 Catch possible `RuntimeException` from `getCameraCharacteristics` by Xavier Lepaul · 3 years, 10 months ago
  31. 11b92cf Refactoring: Move groups-by-mid into Bundle manager by Harald Alvestrand · 3 years, 10 months ago
  32. de22ab2 Apply IWYU to jsep_transport_controller/collection by Harald Alvestrand · 3 years, 10 months ago
  33. d354ced Mark VideoSendTiming flags as invalid by default. by philipel · 3 years, 10 months ago
  34. ada810a Reland "Deprecate microsecond timestamps in RTC event log." by Björn Terelius · 3 years, 10 months ago
  35. 1bb36d2 Change YuvConverter.convert to catch GLExceptions and return null. by Fabian Bergmark · 3 years, 10 months ago
  36. ac82bd3 Add timestamp to log message in generic_decoder.cc by Johannes Kron · 3 years, 10 months ago
  37. 41c700d Remove unnused build configs for M1 builder by Christoffer Jansson · 3 years, 10 months ago
  38. 82f21fd Make WebRtcAudioReceiveStream::stream_ const. by Tommi · 3 years, 10 months ago
  39. b4100ad Avoid using legacy rtp parser in neteq test::Packet by Danil Chapovalov · 3 years, 10 months ago
  40. 35b21ba In RtcpTransceiver avoid extra PostTask during construction by Danil Chapovalov · 3 years, 10 months ago
  41. f9d5e55 Revert "Avoid video stream allocation on configuration change after timeout." by Jakob Ivarsson · 3 years, 10 months ago
  42. a3796c8 Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap. by Christoffer Rodbro · 3 years, 10 months ago
  43. ce3b3ba Update WebRTC code version (2021-06-17T04:05:50). by webrtc-version-updater · 3 years, 10 months ago
  44. 4b62952 Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293) by chromium-webrtc-autoroll · 3 years, 10 months ago
  45. e0c7365 Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176) by chromium-webrtc-autoroll · 3 years, 10 months ago
  46. a2a073b Reformat pc/g3doc/rtp.md by Artem Titov · 3 years, 10 months ago
  47. 55107c8 Update the sync_group id without recreating audio receive streams. by Tommi · 3 years, 10 months ago
  48. 25029c4 Roll chromium_revision b452ca696d..19c2bebe7d (892948:893060) by chromium-webrtc-autoroll · 3 years, 10 months ago
  49. 355c473 Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. by philipel · 3 years, 10 months ago
  50. 5b9d0c7 AGC1 add clipping predictor evaluator by Alessio Bazzica · 3 years, 10 months ago
  51. 808f494 LOG DTLS (failed) handshake retransmission by Jonas Oreland · 3 years, 10 months ago
  52. d579e6b dcsctp: Do explicit bounds checking in bounded IO by Victor Boivie · 3 years, 10 months ago
  53. 72b7998 Remove the `createDecoder(String)` overload by Xavier Lepaul · 3 years, 10 months ago
  54. 130e031 Roll chromium_revision 570a173256..b452ca696d (892156:892948) by chromium-webrtc-autoroll · 3 years, 10 months ago
  55. 98ff028 AGC analog ClippingPredictor refactoring 2/2 by Alessio Bazzica · 3 years, 10 months ago
  56. 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 3 years, 10 months ago
  57. bc03259 Define generate_location_tags gn arg by Björn Terelius · 3 years, 10 months ago
  58. 6a0a559 Reland "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 3 years, 10 months ago
  59. c03d6e9 Support Java_Buffer_toI420 returning null by Fabian Bergmark · 3 years, 10 months ago
  60. cd430c8 Update WebRTC code version (2021-06-16T04:05:58). by webrtc-version-updater · 3 years, 10 months ago
  61. d6957c2 Revert "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 3 years, 10 months ago
  62. e9ae472 Correctly handle retransmissions/padding in early loss detection. by Erik Språng · 3 years, 10 months ago
  63. e3ceb88 Sanitize hostname literals when mDNS obfuscation is on. by Harald Alvestrand · 3 years, 10 months ago
  64. be53049 Reland "Avoid sending empty receiver reports with RtcpTransceiver" by Danil Chapovalov · 3 years, 10 months ago
  65. 7a2db8a Modify Bundle logic to not add & destroy extra transport at add-track by Harald Alvestrand · 3 years, 10 months ago
  66. e4eb8af libstdc++: fix ostream operator<< usage in JsepTransportCollection by Stephan Hartmann · 3 years, 10 months ago
  67. 07bf5b5 Update WebRTC code version (2021-06-15T04:04:38). by webrtc-version-updater · 3 years, 10 months ago
  68. 3008bcd Don't recreate audio receive streams on header extension update. by Tommi · 3 years, 10 months ago
  69. 6bbe1a4 Roll chromium_revision e9261a56ad..570a173256 (892013:892156) by chromium-webrtc-autoroll · 3 years, 10 months ago
  70. d350006 Add rtp_config() accessor to ReceiveStream. by Tommi · 3 years, 10 months ago
  71. 48420fa Revert "Avoid sending empty receiver reports with RtcpTransceiver" by Björn Terelius · 3 years, 10 months ago
  72. 1c1f540 Factor out common receive stream methods to a common interface. by Tommi · 3 years, 10 months ago
  73. e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 3 years, 10 months ago
  74. e5f1a39 Avoid sending empty receiver reports with RtcpTransceiver by Danil Chapovalov · 3 years, 10 months ago
  75. 8b69290 Fix VideoStreamEncoder QP tests to not use SetHasInternalSource by Niels Möller · 3 years, 10 months ago
  76. b237a87 AGC analog ClippingPredictor refactoring 1/2 by Alessio Bazzica · 3 years, 10 months ago
  77. 1ff491b Roll chromium_revision 8907aace7e..e9261a56ad (891631:892013) by chromium-webrtc-autoroll · 3 years, 10 months ago
  78. 74cc9ea Don't register invalid encode complete callbacks. by Peter Hanspers · 3 years, 10 months ago
  79. 1081487 Avoid video stream allocation on configuration change after timeout. by Jakob Ivarsson · 3 years, 11 months ago
  80. 13dac0c Update WebRTC code version (2021-06-13T04:02:18). by webrtc-version-updater · 3 years, 11 months ago
  81. 0f9a8e33 Make stopping of the RepeatingTask safer by Danil Chapovalov · 3 years, 11 months ago
  82. 4bb81ac Make JsepTransportCollection self-managing for transports by Harald Alvestrand · 3 years, 11 months ago
  83. 63c96ce Roll chromium_revision d2f297f391..8907aace7e (890623:891631) by chromium-webrtc-autoroll · 3 years, 11 months ago
  84. 8d3396d In vp9 encoder fuzzer reduce information stored for older frames by Danil Chapovalov · 3 years, 11 months ago
  85. a63d152 AEC3: Unbounded echo spectrum for dominant nearend detection. by Gustaf Ullberg · 3 years, 11 months ago
  86. 1b4807f count webrtc pranswer usage by Philipp Hancke · 3 years, 11 months ago
  87. b22abbc Add kron as owner of api/uma_metrics.h by Johannes Kron · 3 years, 11 months ago
  88. ef4edaf Remove DEPS that was removed from chromium by Andrey Logvin · 3 years, 11 months ago
  89. ec6b655 Break out pc/session_description build target (part 2) by Harald Alvestrand · 3 years, 11 months ago
  90. f5f7e8e Ensure that fps adaptation count can go back to zero when framerate is unrestricted. by Åsa Persson · 3 years, 11 months ago
  91. c63ae48 Prepare for breakout of session_description.{h,cc} by Harald Alvestrand · 3 years, 11 months ago
  92. 62ec0f6 Add small cooldown to unsignalled ssrc stream creation. by Henrik Boström · 3 years, 11 months ago
  93. ba7da8b Relax expectation in OveruseFrameDetectorTest2.ConvergesSlowly by Niels Möller · 3 years, 11 months ago
  94. 9dea393 Move MID/JsepTransport mappings into a new manager object. by Harald Alvestrand · 3 years, 11 months ago
  95. 64e3a36 Update WebRTC code version (2021-06-10T04:05:52). by webrtc-version-updater · 3 years, 11 months ago
  96. 5e65dd5 Add MB configs for M1 bots by Christoffer Jansson · 3 years, 11 months ago
  97. 3cc68ec Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz. by Tommi · 3 years, 11 months ago
  98. e2e0464 Remove a couple of locks from ChannelReceive and add thread checks. by Tommi · 3 years, 11 months ago
  99. b56a63e dcsctp: Prevent overflow of missing parameters by Victor Boivie · 3 years, 11 months ago
  100. 6eda26c Reland "Remove AudioReceiveStream::Reconfigure() method." by Tommi · 3 years, 11 months ago