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  1. e99f687 Move WebRTC to non deprecated jsoncpp APIs. by Mirko Bonadei · 4 years ago main
  2. 06b8f7e Move supported_platforms.md under g3doc/. by Mirko Bonadei · 4 years ago
  3. e6ab520 First draft of WebRTC supported platforms/compilers documentation. by Mirko Bonadei · 4 years ago
  4. 7208457 Same length for all ARM64 platforms Update more audio checksums for M1 by Christoffer Jansson · 4 years ago lkgr
  5. 74543b7 PlatformThreadTest: fix flake. by Markus Handell · 4 years ago
  6. 2b3a10e Add MAC arm64 platform and update checksums for acm unittest by Christoffer Jansson · 4 years ago
  7. 5e82c75 Remove TODOs to remove SetAudioPlayback and SetAudioRecording by Harald Alvestrand · 4 years ago
  8. 482b7c0 Fix -Wimplicit-int-float-conversions. by Peter Kasting · 4 years ago
  9. 64851c0 Reland: Fix echo return loss stats and add to RTCAudioSourceStats. by Taylor Brandstetter · 4 years ago
  10. a987429 Update WebRTC code version (2021-06-25T04:04:09). by webrtc-version-updater · 4 years ago
  11. c830bd6 Remove ssl_certificate() accessor. by Harald Alvestrand · 4 years ago
  12. e9a74c9 Public RtpVideoFrameAssembler by philipel · 4 years ago
  13. 4e51334 AV1 OBU test helper. by philipel · 4 years ago
  14. 28e582d 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 · 4 years ago
  15. f2ed401 Fix unscaled timestamps passed to nack_tracker by Jared Siskin · 4 years ago
  16. 9233af3 Update dependencies on deprecated target rtc_base:critical_section by Niels Möller · 4 years ago
  17. 0742f52 Triggering build after flaky builders (asan). by Tommi · 4 years ago
  18. eb61b7f ModuleRtcRtcpImpl2: remove Module inheritance. by Markus Handell · 4 years ago
  19. 6e65f6a Deprecating AbsoluteCaptureTimeReceiver by Minyue Li · 4 years ago
  20. 3f7b717 RTCPSender: remove compatibility ctor & method. by Markus Handell · 4 years ago
  21. 49cb459 TaskQueueStdlib: initialize the thread last. by Markus Handell · 4 years ago
  22. 0fe60bd Add RecursiveCriticalSection to the don't-use list of primitives by Harald Alvestrand · 4 years ago
  23. c413c55 Replace use of RecursiveCriticalSection in VirtualSocketServer by Niels Möller · 4 years ago
  24. fe6580f Revert "Fix echo return loss stats and add to RTCAudioSourceStats." by Evan Shrubsole · 4 years ago
  25. 9e2b315 Minor code cleanup of WebRtcVideoReceiveStream. by Tommi · 4 years ago
  26. 885d538 ModuleRtpRtcpImpl2: remove RTCP send polling. by Markus Handell · 4 years ago
  27. 2086209 Update WebRTC code version (2021-06-22T04:05:30). by webrtc-version-updater · 4 years ago
  28. 049ed44 ModuleRtpRtcpImpl2: update test code. by Markus Handell · 4 years ago
  29. 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 · 4 years ago
  30. c6b9ac7 RTCPSender: migrate to Timestamp. by Markus Handell · 4 years ago
  31. e2ab77b Reland "Port: migrate to TaskQueue." by Markus Handell · 4 years ago
  32. a27cfbf Fix echo return loss stats and add to RTCAudioSourceStats. by Taylor Brandstetter · 4 years ago
  33. 2e3edc1 RTCPSender: migrate to own configuration struct. by Markus Handell · 4 years ago
  34. f906ec4 Handle null return from ToI420 in encoders by Evan Shrubsole · 4 years ago
  35. 76a35d9 Delete legacy RtpHeaderParser wrapper by Danil Chapovalov · 4 years ago
  36. 257f81b Update VirtualSocketServer locking to match documentation. by Niels Möller · 4 years ago
  37. a4aabb9 Revert "Port: migrate to TaskQueue." by Markus Handell · 4 years ago
  38. 0654016 Port: migrate to TaskQueue. by Markus Handell · 4 years ago
  39. 3f6efdc Update WebRTC code version (2021-06-21T04:05:45). by webrtc-version-updater · 4 years ago
  40. ae278d4 openssl_adapter: document SSL_CTX_set_verify_depth behaviour by Philipp Hancke · 4 years ago
  41. fbe9958 Update WebRTC code version (2021-06-20T04:03:02). by webrtc-version-updater · 4 years ago
  42. 4cacaf7 Update WebRTC code version (2021-06-19T04:03:03). by webrtc-version-updater · 4 years ago
  43. 7c719b0 Fixes off-by-one error in video capture module by Johannes Kron · 4 years ago
  44. bad0ab0 Delete unused class MockDelayable by Niels Möller · 4 years ago
  45. c6d7648 Add jakobi to modules/audio_coding OWNERS by Ivo Creusen · 4 years ago
  46. 6a11c84 dcsctp: Add DcSctpSocketFactory by Florent Castelli · 4 years ago
  47. c20f156 dcsctp: Don't sent more packets before COOKIE ACK by Victor Boivie · 4 years ago
  48. 95c3041 Update WebRTC code version (2021-06-18T04:03:27). by webrtc-version-updater · 4 years ago
  49. 42dacda AGC analog clipping predictor: integrate evaluator by Alessio Bazzica · 4 years ago
  50. 7d54182 Avoid assembling complicated but unused video rtp header extensions by Danil Chapovalov · 4 years ago
  51. afb2811 Catch possible `RuntimeException` from `getCameraCharacteristics` by Xavier Lepaul · 4 years ago
  52. 11b92cf Refactoring: Move groups-by-mid into Bundle manager by Harald Alvestrand · 4 years ago
  53. de22ab2 Apply IWYU to jsep_transport_controller/collection by Harald Alvestrand · 4 years ago
  54. d354ced Mark VideoSendTiming flags as invalid by default. by philipel · 4 years ago
  55. ada810a Reland "Deprecate microsecond timestamps in RTC event log." by Björn Terelius · 4 years ago
  56. 1bb36d2 Change YuvConverter.convert to catch GLExceptions and return null. by Fabian Bergmark · 4 years ago
  57. ac82bd3 Add timestamp to log message in generic_decoder.cc by Johannes Kron · 4 years ago
  58. 41c700d Remove unnused build configs for M1 builder by Christoffer Jansson · 4 years ago
  59. 82f21fd Make WebRtcAudioReceiveStream::stream_ const. by Tommi · 4 years ago
  60. b4100ad Avoid using legacy rtp parser in neteq test::Packet by Danil Chapovalov · 4 years ago
  61. 35b21ba In RtcpTransceiver avoid extra PostTask during construction by Danil Chapovalov · 4 years ago
  62. f9d5e55 Revert "Avoid video stream allocation on configuration change after timeout." by Jakob Ivarsson · 4 years ago
  63. a3796c8 Revert the send-side bwe behavior to slow ramp-up on lifted REMB cap. by Christoffer Rodbro · 4 years ago
  64. ce3b3ba Update WebRTC code version (2021-06-17T04:05:50). by webrtc-version-updater · 4 years ago
  65. 4b62952 Roll chromium_revision 6ade74989a..6f7025c98c (893176:893293) by chromium-webrtc-autoroll · 4 years ago
  66. e0c7365 Roll chromium_revision 19c2bebe7d..6ade74989a (893060:893176) by chromium-webrtc-autoroll · 4 years ago
  67. a2a073b Reformat pc/g3doc/rtp.md by Artem Titov · 4 years ago
  68. 55107c8 Update the sync_group id without recreating audio receive streams. by Tommi · 4 years ago
  69. 25029c4 Roll chromium_revision b452ca696d..19c2bebe7d (892948:893060) by chromium-webrtc-autoroll · 4 years ago
  70. 355c473 Fix VideoRtpDepacketizerVp{8,9} copy assignment signature. by philipel · 4 years ago
  71. 5b9d0c7 AGC1 add clipping predictor evaluator by Alessio Bazzica · 4 years ago
  72. 808f494 LOG DTLS (failed) handshake retransmission by Jonas Oreland · 4 years ago
  73. d579e6b dcsctp: Do explicit bounds checking in bounded IO by Victor Boivie · 4 years ago
  74. 72b7998 Remove the `createDecoder(String)` overload by Xavier Lepaul · 4 years ago
  75. 130e031 Roll chromium_revision 570a173256..b452ca696d (892156:892948) by chromium-webrtc-autoroll · 4 years ago
  76. 98ff028 AGC analog ClippingPredictor refactoring 2/2 by Alessio Bazzica · 4 years ago
  77. 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 4 years ago
  78. bc03259 Define generate_location_tags gn arg by Björn Terelius · 4 years ago
  79. 6a0a559 Reland "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 4 years ago
  80. c03d6e9 Support Java_Buffer_toI420 returning null by Fabian Bergmark · 4 years ago
  81. cd430c8 Update WebRTC code version (2021-06-16T04:05:58). by webrtc-version-updater · 4 years ago
  82. d6957c2 Revert "Correctly handle retransmissions/padding in early loss detection." by Erik Språng · 4 years ago
  83. e9ae472 Correctly handle retransmissions/padding in early loss detection. by Erik Språng · 4 years ago
  84. e3ceb88 Sanitize hostname literals when mDNS obfuscation is on. by Harald Alvestrand · 4 years ago
  85. be53049 Reland "Avoid sending empty receiver reports with RtcpTransceiver" by Danil Chapovalov · 4 years ago
  86. 7a2db8a Modify Bundle logic to not add & destroy extra transport at add-track by Harald Alvestrand · 4 years ago
  87. e4eb8af libstdc++: fix ostream operator<< usage in JsepTransportCollection by Stephan Hartmann · 4 years ago
  88. 07bf5b5 Update WebRTC code version (2021-06-15T04:04:38). by webrtc-version-updater · 4 years ago
  89. 3008bcd Don't recreate audio receive streams on header extension update. by Tommi · 4 years ago
  90. 6bbe1a4 Roll chromium_revision e9261a56ad..570a173256 (892013:892156) by chromium-webrtc-autoroll · 4 years ago
  91. d350006 Add rtp_config() accessor to ReceiveStream. by Tommi · 4 years ago
  92. 48420fa Revert "Avoid sending empty receiver reports with RtcpTransceiver" by Björn Terelius · 4 years ago
  93. 1c1f540 Factor out common receive stream methods to a common interface. by Tommi · 4 years ago
  94. e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 4 years ago
  95. e5f1a39 Avoid sending empty receiver reports with RtcpTransceiver by Danil Chapovalov · 4 years ago
  96. 8b69290 Fix VideoStreamEncoder QP tests to not use SetHasInternalSource by Niels Möller · 4 years ago
  97. b237a87 AGC analog ClippingPredictor refactoring 1/2 by Alessio Bazzica · 4 years ago
  98. 1ff491b Roll chromium_revision 8907aace7e..e9261a56ad (891631:892013) by chromium-webrtc-autoroll · 4 years ago
  99. 74cc9ea Don't register invalid encode complete callbacks. by Peter Hanspers · 4 years ago
  100. 1081487 Avoid video stream allocation on configuration change after timeout. by Jakob Ivarsson · 4 years ago