1. 625e1d9 VP8 ref finder unittest cleanup by philipel · 3 hours ago master
  2. 86f04ad Populate “fractionLost” stats for remote inbound rtp streams by Di Wu · 4 hours ago
  3. 6512af0 Add root doc site definition for WebRTC documentation by Artem Titov · 4 hours ago
  4. f57f2cd doc: add M88/M89 release notes by Philipp Hancke · 4 hours ago
  5. 64f7da0 Use input_state to get pixels for single active stream. by Åsa Persson · 5 hours ago
  6. 31e06cb addIceCandidate: prefer ice candidate sdpMid over sdpMLineIndex by Philipp Hancke · 7 hours ago
  7. df6d4ca Prevent TURN from connecting to ports < 1024 (except 443) by Harald Alvestrand · 7 hours ago
  8. bfc3ef0 Mention ninja all in webrtc development documantation. by Andrey Logvin · 8 hours ago
  9. 4785402 Replace RecursiveCriticalSection with Mutex in ProcessThreadImpl by Niels Möller · 8 hours ago
  10. 1e2da37 Update WebRTC code version (2021-02-28T04:02:30). by webrtc-version-updater · 2 days ago
  11. ac615b6 Update WebRTC code version (2021-02-27T04:03:14). by webrtc-version-updater · 3 days ago
  12. c96aa30 Minor refactoring of RtpVideoSender by Erik Språng · 3 days ago
  13. 484acf2 Add ability to configure sampling rate for input/output video dumps in PC level framework by Artem Titov · 3 days ago
  14. 9d9b8de Reland "Rename SIGNALING and WORKER to PRIMARY and SECONDARY" by Mirko Bonadei · 3 days ago
  15. 376cf38 Replace RecursiveCriticalSection with Mutex in EmulatedEndpointImpl by Niels Möller · 3 days ago lkgr
  16. 07a01d0 Revert "Rename SIGNALING and WORKER to PRIMARY and SECONDARY" by Mirko Bonadei · 4 days ago
  17. a37f2bd Rename SIGNALING and WORKER to PRIMARY and SECONDARY by Harald Alvestrand · 4 days ago
  18. 2aeab5e Make the PC proxy invoke LookupDtlsTransportByMid on the network thread by Tomas Gunnarsson · 4 days ago
  19. 5cfcf22 modules/desktop_capture: replace memcpy with libyuv::CopyPlane by Zhaoliang Ma · 4 days ago
  20. 258e989 Use default ResolutionBitrateLimits for simulcast with one active stream if not configured by Åsa Persson · 4 days ago
  21. 1124ed1 Communicate encoder resolutions via rtc::VideoSinkWants. by Henrik Boström · 4 days ago
  22. bb52bdf Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium" by Johannes Kron · 4 days ago
  23. cd5127b Revert "Enable use of rtc::SystemTimeNanos() provided by Chromium" by Mirko Bonadei · 4 days ago
  24. 49b2792 Update WebRTC code version (2021-02-25T04:03:13). by webrtc-version-updater · 5 days ago
  25. c500977 Change the safe SCTP MTU size to 1191 by Tomas Gunnarsson · 5 days ago
  26. dfe1971 Enable use of rtc::SystemTimeNanos() provided by Chromium by Johannes Kron · 5 days ago
  27. dac39c5 Reland "Add test for odd sizes with spatial layers" by Sergio Garcia Murillo · 5 days ago
  28. 61d1773 Remove deactivated RTP modules from PacketRouter map. by Erik Språng · 5 days ago
  29. 451a8af Feed the clock skew to AbsoluteCaptureTimeReceiver in audio receiver. by Minyue Li · 5 days ago
  30. cd0373f peerconnection: add was-ever-connected boolean flag by Philipp Hancke · 5 days ago
  31. 2ee9415 AndroidVideoDecoder: Ignore format updates with zero dimensions by Raman Budny · 5 days ago
  32. eaedde7 Remove old workaround in PacingController by Erik Språng · 5 days ago
  33. 0093a38 Fix low-latency renderer with unset playout delays by Johannes Kron · 5 days ago
  34. 198299c Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits) by Keiichi Enomoto · 5 days ago
  35. 77475ec Delete unused sigslot variables. by Lahiru Ginnaliya Gamathige · 6 days ago
  36. c32f00e Remove usage of sigslot and replace with a function pointer. by Lahiru Ginnaliya Gamathige · 6 days ago
  37. 5fec23c Update WebRTC code version (2021-02-24T04:02:50). by webrtc-version-updater · 6 days ago
  38. 8af6b49 Populate jitter stats for video RTP streams by Di Wu (RP Room Eng) · 6 days ago
  39. 373bb7b Don't use SystemTimeNanos() for current wallclock time on WINUWP by Johannes Kron · 6 days ago
  40. d0844a8 Revert "Add test for odd sizes with spatial layers" by Florent Castelli · 6 days ago
  41. 09226fc Disable high-pass filtering of the AEC reference by Gustaf Ullberg · 7 days ago
  42. e5caa9e Update WebRTC code version (2021-02-23T04:03:09). by webrtc-version-updater · 7 days ago
  43. 90ea0a6 AV1: Change multithreading, speed, qp settings by Jerome Jiang · 7 days ago
  44. 6fe3fa1 Add test for odd sizes with spatial layers by Sergio Garcia Murillo · 7 days ago
  45. 28547e9 Fix typos in network emulation default routing by Artem Titov · 7 days ago
  46. d44532a Delete unused sigslot SignalAddressReady and MSG_ID_ADDRESS_BOUND by Niels Möller · 7 days ago
  47. 77ee854 Extract sequencing from RtpSender by Erik Språng · 7 days ago
  48. 7013b3b Add deprecation section to webrtc style guide by Danil Chapovalov · 7 days ago
  49. e904161 Replace RTC_DEPRECATED with ABSL_DEPRECATED by Danil Chapovalov · 7 days ago
  50. 8ef1d7b Add a missing lock in VideoBroadcaster::OnDiscardedFrame(). by Mirta Dvornicic · 7 days ago
  51. 2bfddf7 Add thread annotations and docs in ProcessThreadImpl. by Niels Möller · 7 days ago
  52. bc9dc5a Upload all values instead of only mean and err into histograms by Artem Titov · 7 days ago
  53. 3d37e06 Introduce default routes for network emulation by Artem Titov · 7 days ago
  54. 1dd94a0 Use pixels from single active stream if set in CanDecreaseResolutionTo by Åsa Persson · 7 days ago
  55. 42dd9bc Add documentation about DefaultVideoQualityAnalyzer by Artem Titov · 7 days ago
  56. a21ea29 Update WebRTC code version (2021-02-20T04:03:37). by webrtc-version-updater · 10 days ago
  57. 04a6529 AV1: set superblock to 64x64 for 720p 4 threads. by Jerome Jiang · 10 days ago
  58. 1fbff10 In RtpVideoStreamReceiver change way to track time for the last received packet. by Danil Chapovalov · 10 days ago
  59. f3dc47e Ending a statement with a semicolon by Zhaoliang Ma · 10 days ago
  60. da20c73 Add build argument rtc_exclude_system_time by Johannes Kron · 10 days ago
  61. 072c008 Reland "Replace RecursiveCriticalSection with Mutex in RTCAudioSession." by Niels Möller · 10 days ago
  62. ae096ef Remove log message if balanced/cpu speed field trial is not set. by Åsa Persson · 10 days ago
  63. 16359f6 Delay creation of decoders until they are needed by Johannes Kron · 10 days ago
  64. c9b9930 Add L2T3 K-SVC structure by Emil Lundmark · 10 days ago
  65. 753c76a Update WebRTC code version (2021-02-19T04:02:42). by webrtc-version-updater · 11 days ago
  66. 735e33f Add S3T3 video scalability structure by Danil Chapovalov · 11 days ago
  67. 0f71871 Reland "Batch assign RTP seq# for all packets of a frame." by Erik Språng · 11 days ago
  68. 9915db3 Move Call's histogram reporting code into destructor. by Tomas Gunnarsson · 11 days ago
  69. 17f914c Revert "Batch assign RTP seq# for all packets of a frame." by Jeremy Leconte · 11 days ago
  70. e11b4ae doc: show how to build the fuzzers by Philipp Hancke · 12 days ago
  71. 86d3725 Update WebRTC code version (2021-02-18T04:03:24). by webrtc-version-updater · 12 days ago
  72. 5cc9957 Batch assign RTP seq# for all packets of a frame. by Erik Språng · 12 days ago
  73. e927c0f QualityScalingTests: Move encoder factory creation to ScalingObserver. by Åsa Persson · 12 days ago
  74. 3999384 Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" by Harald Alvestrand · 12 days ago
  75. 62b6c92 Refactor LossBasedBandwidthEstimation by Per Kjellander · 12 days ago
  76. ebc563e Update the call to RuntimeEnvironment.application by Artem Titov · 12 days ago
  77. 89c40e2 Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" by Harald Alvestrand · 12 days ago
  78. 60c0b44 Use CallbackList for DtlsState in dtls_transport. by Lahiru Ginnaliya Gamathige · 13 days ago
  79. 46e5a2f Update WebRTC code version (2021-02-17T04:02:09). by webrtc-version-updater · 13 days ago
  80. 772066b Reland "Split peer_connection_integrationtest.cc into pieces" by Harald Alvestrand · 13 days ago
  81. 3562318 Delete unused functions in RtpSender, RtcpSender and RtcpReceiver by Danil Chapovalov · 13 days ago
  82. f4e3e2b Delete rtc::Callback0 and friends. by Niels Möller · 13 days ago
  83. d6c81db Replace VideoLayerFrameId with int64_t. by philipel · 13 days ago
  84. bdf78cb Bug fixes to EglBase10Impl.getNativeEglContext. by Sami Kalliomäki · 13 days ago
  85. 067b050 Delete deprecated unused functions from RtpRtcp interface by Danil Chapovalov · 13 days ago
  86. 2c8d929 QualityScalingTests: Add tests for VP9. by Åsa Persson · 13 days ago
  87. c79bd43 Delete friendship between VirtualSocket and VirtualSocketServer by Niels Möller · 13 days ago
  88. 686ad4f Resolve relative paths in sdk build scripts. by Yura Yaroshevich · 14 days ago
  89. d1bbec3 Update WebRTC code version (2021-02-16T04:03:42). by webrtc-version-updater · 14 days ago
  90. b73c9f0 Extract SystemTimeNanos to its own file by Johannes Kron · 14 days ago
  91. 8408c99 Remove 'secondary sink' concept from webrtc::VideoReceiveStream. by Tomas Gunnarsson · 2 weeks ago
  92. a33f41b Support getNativeEglContext in EglBase10Impl. by Sami Kalliomäki · 2 weeks ago
  93. 8623c75 Remove ctor for BuiltInNetworkBehaviorConfig by Per Kjellander · 2 weeks ago
  94. 453a125 Remove no longer needed FrameDroppingOn setting in QualityScalingTests. by Åsa Persson · 2 weeks ago
  95. 51746ce Revert "Replace RecursiveCriticalSection with Mutex in RTCAudioSession." by Niels Moller · 2 weeks ago
  96. 17ec2fc Remove log line that states that FlexFEC is disabled. by Rasmus Brandt · 2 weeks ago
  97. 6e35ece Destroy PC properly to stop input video before closing video writer by Artem Titov · 2 weeks ago
  98. 9aa9b8d Prepare to replace VideoLayerFrameId with int64_t. by philipel · 2 weeks ago
  99. 563fbc1 Replace RecursiveCriticalSection with Mutex in DxgiDuplicatorController by Niels Möller · 2 weeks ago
  100. 410c998 Const correct NetworkEmulationManager::GetStats by Per Kjellander · 2 weeks ago