1. 8679ac1 Fix unsignalled ssrc race in WebRtcVideoChannel. by Henrik Boström · 3 years, 9 months ago
  2. 006206d rtx-time implementation by Philipp Hancke · 3 years, 10 months ago
  3. 0b5ec18 Simplify ChannelManager initialization. by Tomas Gunnarsson · 3 years, 9 months ago
  4. b258c56 Send and Receive VideoFrameTrackingid RTP header extension. by Jeremy Leconte · 3 years, 10 months ago
  5. 128faf8 Delete AsyncInvoker usage from SctpDataSender by Niels Möller · 3 years, 10 months ago
  6. f7b1b95 Add `RTCRemoteOutboundRtpStreamStats` for audio streams by Alessio Bazzica · 3 years, 10 months ago
  7. 7cbe887 Change default adaptive ptime min bitrate to 16kbps. by Jakob Ivarsson · 3 years, 10 months ago
  8. c780605 Make num_encoded_channels_ atomic by Gustaf Ullberg · 3 years, 10 months ago
  9. cf93670 sctp: Finish sending partial messages before sending stream reset events by Florent Castelli · 3 years, 10 months ago
  10. d19e3b9 Reland "Reland "Enable quality scaling when allowed"" by Sergey Silkin · 3 years, 10 months ago
  11. 31c5c9d Revert "Reland "Enable quality scaling when allowed"" by Ilya Nikolaevskiy · 3 years, 10 months ago
  12. 0021fe7 Reland "Enable quality scaling when allowed" by Sergey Silkin · 3 years, 10 months ago
  13. b2e71b8 Reland "Fix race between destroying SctpTransport and receiving notification on timer thread." by Taylor Brandstetter · 3 years, 10 months ago
  14. 668dbf6 [Stats] Populate "frames" stats for video source. by Di Wu · 3 years, 11 months ago
  15. eb449a9 Revert "Reland "Enable quality scaling when allowed"" by Guido Urdaneta · 3 years, 10 months ago
  16. 8a38b1c Revert "Fix race between destroying SctpTransport and receiving notification on timer thread." by Taylor · 3 years, 10 months ago
  17. 83be84b Reland "Enable quality scaling when allowed" by Sergey Silkin · 3 years, 10 months ago
  18. 609b524 Revert "Enable quality scaling when allowed" by Björn Terelius · 3 years, 10 months ago
  19. 752cbab Enable quality scaling when allowed by Sergey Silkin · 3 years, 10 months ago
  20. a88fe7b Fix race between destroying SctpTransport and receiving notification on timer thread. by Taylor Brandstetter · 3 years, 11 months ago
  21. c500977 Change the safe SCTP MTU size to 1191 by Tomas Gunnarsson · 3 years, 11 months ago
  22. 8af6b49 Populate jitter stats for video RTP streams by Di Wu (RP Room Eng) · 3 years, 11 months ago
  23. e904161 Replace RTC_DEPRECATED with ABSL_DEPRECATED by Danil Chapovalov · 3 years, 11 months ago
  24. 8ef1d7b Add a missing lock in VideoBroadcaster::OnDiscardedFrame(). by Mirta Dvornicic · 3 years, 11 months ago
  25. f4e3e2b Delete rtc::Callback0 and friends. by Niels Möller · 4 years ago
  26. 8408c99 Remove 'secondary sink' concept from webrtc::VideoReceiveStream. by Tomas Gunnarsson · 4 years ago
  27. 17ec2fc Remove log line that states that FlexFEC is disabled. by Rasmus Brandt · 3 years, 11 months ago
  28. 2afff37 Update field trial for allowing cropped resolution when limiting max layers. by Åsa Persson · 4 years ago
  29. e71b55f build: merge media_constants and engine_constants by Philipp Hancke · 4 years ago
  30. d15a575 Use SequenceChecker from public API by Artem Titov · 4 years ago
  31. ad32586 Reland "Prepare to avoid hops to worker for network events." by Tomas Gunnarsson · 4 years ago
  32. 47ec157 Revert "Prepare to avoid hops to worker for network events." by Mirko Bonadei · 4 years ago
  33. 0e3cb9f Create and initialize encoders only for active streams by Sergey Silkin · 4 years ago
  34. d48a2b1 Prepare to avoid hops to worker for network events. by Tomas Gunnarsson · 4 years ago
  35. c8421c4 Replace rtc::ThreadChecker with webrtc::SequenceChecker by Artem Titov · 4 years ago
  36. e7c79fd Remove from chromium build targets that are not compatible with it. by Andrey Logvin · 4 years ago
  37. 7864600 Add absl_deps field for rtc_test and rtc_executable by Andrey Logvin · 4 years ago
  38. 9673ca4 Add field trial for bitrate limit interpolation for simulcast resolutions <180p. by Rasmus Brandt · 4 years ago
  39. a7e34d3 Add resolution_bitrate_limits to EncoderInfo field trial. by Åsa Persson · 4 years ago
  40. 49b20f9 Fix race with SctpTransport destruction and usrsctp timer thread. by Taylor Brandstetter · 4 years ago
  41. 33c0ab4 Call MediaChannel::OnPacketReceived on the network thread. by Tomas Gunnarsson · 4 years ago
  42. 29bd863 Add field trial for allowing cropped resolution when limiting max layers. by Åsa Persson · 4 years ago
  43. 77ceff9 payload type mapper: use media constants by Philipp Hancke · 4 years ago
  44. c20e333 Update SCTP test to use C++ lambdas instead of rtc::Bind by Niels Möller · 4 years ago
  45. 8467cf2 Reduce redundant flags for audio stream playout state. by Tomas Gunnarsson · 4 years ago
  46. 2397b6e SimulcastEncoderAdapter: Add field trial for EncoderInfo settings. by Åsa Persson · 4 years ago
  47. e5f4c6b Reland "Refactor rtc_base build targets." by Mirko Bonadei · 4 years ago
  48. 5ab6a8c Refactors SimulcastEncoder Adapter. by Erik Språng · 4 years ago
  49. 7acc2d9 Revert "Refactor rtc_base build targets." by Mirko Bonadei · 4 years ago
  50. 42c0d70 Include packetization in video codec string by Emil Lundmark · 4 years ago
  51. ece6712 Add av1 to lower range IDs. by Jerome Jiang · 4 years ago
  52. 69241a9 Refactor rtc_base build targets. by Mirko Bonadei · 4 years ago
  53. e15dc58 Use rtc::CopyOnWriteBuffer::MutableData through webrtc by Danil Chapovalov · 4 years ago
  54. 942976e Wire scalability_mode when simulcast is not in use (i.e. streams==1) by Sergio Garcia Murillo · 4 years ago
  55. b8f32c4 video_engine: fix logging by Philipp Hancke · 4 years ago
  56. b03b6c8 Move setting of encoder bitrate allocation callback type to VideoSendStream by Per Kjellander · 4 years ago
  57. 167ecc9 Use the correct function name in the RTC log output. by Hua, Chunbo · 4 years ago
  58. 7aeb195 flexfec: improve readability by Philipp Hancke · 4 years, 1 month ago
  59. 1e98f95 sdp: remove some unused x-google attributes by Philipp Hancke · 4 years, 1 month ago
  60. 08d2c2b Delete unneeded dependencies on the Module abstraction by Niels Möller · 4 years, 1 month ago
  61. 46ea5d7 Surface the number of encoded channels by Gustaf Ullberg · 4 years, 1 month ago
  62. 064be38 Reland "Enable FlexFEC as a receiver video codec by default" by Harsh Maniar · 4 years, 1 month ago
  63. 04ed0a0 Change LS_ERROR to LS_WARNING for unsupported decoder formats by Johannes Kron · 4 years, 1 month ago
  64. 5c4c836 use [35,65] rtp payload type range for new codecs by Philipp Hancke · 4 years, 1 month ago
  65. 09ca9a1 allow dynamic payload types <= 95 by Philipp Hancke · 4 years, 1 month ago
  66. 6562109 test: do not consider flexfec-03 a normal codec by Philipp Hancke · 4 years, 2 months ago
  67. d381194 Adjust min bitrate for the first active stream by Ilya Nikolaevskiy · 4 years, 2 months ago
  68. b7d89ca Move iOS noise suppression override to default settings by Sam Zackrisson · 4 years, 2 months ago
  69. 3f77eb4 fixing build for when HAVE_WEBRTC_VIDEO is not defined by Daniel Morilha · 4 years, 2 months ago
  70. 47a03e8 Default enable sending transport sequence numbers on audio packets. by Jakob Ivarsson · 4 years, 2 months ago
  71. 20e4c80 Reland "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 2 months ago
  72. ce4be1e Revert "Enable FlexFEC as a receiver video codec by default" by Mirko Bonadei · 4 years, 2 months ago
  73. bff717e Remove dependency on AsyncInvoker in SctpTransport by Tomas Gunnarsson · 4 years, 2 months ago
  74. f08db1b Enable FlexFEC as a receiver video codec by default by Harsh Maniar · 4 years, 2 months ago
  75. 87e9909 Make video scalability mode configurable from peerconnection level. by philipel · 4 years, 2 months ago
  76. 4d18eea Remove resolution limited max bitrate for simulcast screenshare. by Jakob Ivarsson · 4 years, 2 months ago
  77. 1bd6cc5 Make SEA to be codec agnostic. by Sergey Silkin · 4 years, 2 months ago
  78. c95b939 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() by Karl Wiberg · 4 years, 2 months ago
  79. 37aaabb Free data if SCTP packet is delivered after transport destruction. by Taylor Brandstetter · 4 years, 2 months ago
  80. f8b5bfe Fix "control reaches end of non-void function" warnings by Fabien Vallée · 4 years, 3 months ago
  81. afee708 do not set rtp datachannel b=AS for SCTP by Philipp Hancke · 4 years, 3 months ago
  82. 4182b49 Avoid duplicate usrsctp_init if the last usrsctp_finish failed. by Taylor Brandstetter · 4 years, 3 months ago
  83. 70c8945 Offer VideoLayersAllocation if field trial enabled by Per Kjellander · 4 years, 3 months ago
  84. 9f4859e Allow to set av1 scalability mode after encoder is constructed by Danil Chapovalov · 4 years, 3 months ago
  85. 088a6f2 Fix possible deadlock when handling SCTP_SEND_FAILED_EVENT notification. by Taylor Brandstetter · 4 years, 3 months ago
  86. 9da1b8f Distinguish id for GFD and DD rtp header extensions when both are advertised by Danil Chapovalov · 4 years, 3 months ago
  87. 44f749d Advertise dependency descriptor rtp header extension behind a field trial by Danil Chapovalov · 4 years, 3 months ago
  88. 5089a8e Use VideoFrameBuffer::Scale in encoder wrappers by Evan Shrubsole · 4 years, 3 months ago
  89. 38e9b06 Reland "Add scaling interface to VideoFrameBuffer" by Ilya Nikolaevskiy · 4 years, 3 months ago
  90. 441dbf9 Revert "Add scaling interface to VideoFrameBuffer" by Ilya Nikolaevskiy · 4 years, 3 months ago
  91. c79f1d8 Add scaling interface to VideoFrameBuffer by Ilya Nikolaevskiy · 4 years, 3 months ago
  92. dcef641 Stop using VideoBitrateAllocationObserver in VideoStreamEncoder. by Per Kjellander · 4 years, 3 months ago
  93. 0abd518 Revert "Introduce RTC_NO_UNIQUE_ADDRESS." by Mirko Bonadei · 4 years, 3 months ago
  94. c401923 Take max bitrate into account for target bitrate decision when min bitrate is empty by Yun Zhang · 4 years, 3 months ago
  95. 09ceed2 Async audio processing API by Olga Sharonova · 4 years, 3 months ago
  96. f5e261a Introduce RTC_NO_UNIQUE_ADDRESS. by Mirko Bonadei · 4 years, 3 months ago
  97. de95329 Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS by Niels Möller · 4 years, 3 months ago
  98. 9def3ff Fix for OnSctpInboundPacket being called after transport destruction. by Taylor Brandstetter · 4 years, 4 months ago
  99. 7a76835 Check length before dereferencing SCTP notifications. by Taylor Brandstetter · 4 years, 4 months ago
  100. ceb4495 Reland: Wires up WebrtcKeyValueBasedConfig in media engines. by Erik Språng · 4 years, 4 months ago