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webrtc / src.git / 787e326c180c4b2704c7ede99fa335b4fe4a5e83 / media / engine
  1. 875cd32 Fix inconsistency with x-goog-max-bitrate and maxBitrate. by Henrik Boström · 1 year, 7 months ago
  2. b90cd91 Fix first encoding's maxBitrate being ignored when scalability is set. by Henrik Boström · 1 year, 7 months ago
  3. 0145db4 Recreate the stream when switching from standard to legacy API. by Henrik Boström · 1 year, 7 months ago
  4. e2dff95 Revert "Clean up WebRTC-FilterAbsSendTimeExtension field trial" by Mirko Bonadei · 1 year, 8 months ago
  5. ebf7111 Clean up WebRTC-FilterAbsSendTimeExtension field trial by Philipp Hancke · 1 year, 8 months ago
  6. cabd77a Remove flexfec-03 killswitch guarding receiving FlexFEC by Philipp Hancke · 1 year, 8 months ago
  7. d797cb6 Remove all split channels related code by Florent Castelli · 1 year, 8 months ago
  8. 96293f0 Remove usage of CreateMediaChannel in webrtc_voice_engine_unittest by Florent Castelli · 1 year, 8 months ago
  9. 0776415 Generalize stream parameter primary/secondary ssrc checks by Philipp Hancke · 1 year, 8 months ago
  10. 84fdf99 Convert Media*Channel to contain a webrtc::Transport by Harald Alvestrand · 1 year, 9 months ago
  11. d20bbc4 Remove CreateMediaChannel calls from webrtc_video_engine_unittest by Florent Castelli · 1 year, 9 months ago
  12. 17e8a5c stats: implement flexfec fecBytesReceived stats for FlexFEC by Philipp Hancke · 1 year, 9 months ago
  13. 4e434c3 Remove MediaChannel usage from webrtc_video_engine_unittest by Florent Castelli · 1 year, 9 months ago
  14. ee97e6a Move GetSendCodec() to MediaSendChannelInterface by Florent Castelli · 1 year, 9 months ago
  15. 328e7b2 Sort media/engine/webrtc_video_engine.cc by Harald Alvestrand · 1 year, 9 months ago
  16. 213090b Add AbsoluteCaptureTime RTP extension to supported list in engines. by Florent Castelli · 1 year, 9 months ago
  17. 1cb54be Delete unused killswitch flag related to scalability mode. by Henrik Boström · 1 year, 9 months ago
  18. c0e2418 Sort WebRtcAudio{Send,Receive}Channel implementation by Harald Alvestrand · 1 year, 9 months ago
  19. 682755e Do not support frame tracking id extension in production by Philipp Hancke · 1 year, 10 months ago
  20. 09e0086 Remove ImplForTesting function from MediaChannel by Harald Alvestrand · 1 year, 9 months ago
  21. 847208e Remove transitional shim classes by Harald Alvestrand · 1 year, 9 months ago
  22. 8c4b9ea Remove references to AudioCodec and VideoCodec constructors by Florent Castelli · 1 year, 9 months ago
  23. 77c6230 Add create functions for voice media send and receive channels. by Harald Alvestrand · 1 year, 9 months ago
  24. b0ef5e4 Declare factory functions for video sender and receiver by Harald Alvestrand · 1 year, 9 months ago
  25. 2f0c078 Split WebRtcVoiceChannel into Send and Receive classes by Harald Alvestrand · 1 year, 9 months ago
  26. 811e24a Move functionality from AudioCodec and VideoCodec into cricket::Codec by Florent Castelli · 1 year, 9 months ago
  27. 54e95bc Propagate time of the last received packet with Timestamp type by Danil Chapovalov · 1 year, 9 months ago
  28. 9a34d80 Apply the "shim" pattern for WebRtcVoiceEngine by Harald Alvestrand · 1 year, 9 months ago
  29. f785bd4 Split WebRtcVideoMediaChannel into Send and Receive by Harald Alvestrand · 1 year, 9 months ago
  30. 4ad141e Add callback for send codec in audio too by Harald Alvestrand · 1 year, 9 months ago
  31. a9bba04 Updating AsyncAudioProcessing API, part 1. by Peter Hanspers · 1 year, 9 months ago
  32. d8b88d8 Use the VideoMediaChannelShim for all cases by Harald Alvestrand · 1 year, 9 months ago
  33. 97c9623 Make a shim object implementing the VideoMediaChannel interface by Harald Alvestrand · 1 year, 9 months ago
  34. f0820ff Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay by Rasmus Brandt · 1 year, 10 months ago
  35. 5f32fa4 Delete MediaBaseChannel class by Harald Alvestrand · 1 year, 9 months ago
  36. cfd4cd0 Introduce AddDefaultRecvStreamForTesting to VideoReceiveChannel API by Harald Alvestrand · 1 year, 10 months ago
  37. 621cb29 Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec. by Rasmus Brandt · 1 year, 10 months ago
  38. 4858a0d Add test for split-mode SSRC callback by Harald Alvestrand · 1 year, 10 months ago
  39. 13897e6 Change SSRC-passing for MediaChannel from external to callback by Harald Alvestrand · 1 year, 10 months ago
  40. 487c943 Guard send_codec variable against receive channel access by Harald Alvestrand · 1 year, 10 months ago
  41. 7924915 Stop decoding video for m-lines which are sendonly or inactive by Philipp Hancke · 1 year, 10 months ago
  42. 63551c6 Initialize RTP modes from callback by Harald Alvestrand · 1 year, 10 months ago
  43. e90d934 Delete the WebRTC-H264Simulcast/Disabled/ field trial. by Henrik Boström · 1 year, 10 months ago
  44. e32b622 RtpTransportControllerSend::ProcessSentPacket: remove PostTask. by Markus Handell · 1 year, 10 months ago
  45. c8c4a28 Introduce support for video packet batching. by Markus Handell · 1 year, 10 months ago
  46. ea33f7f Cleanup usasge of ReportBlockData::report_block accessor by Danil Chapovalov · 1 year, 10 months ago
  47. bceec84 Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ by Jared Siskin · 1 year, 11 months ago
  48. f78d1f2 stats: Implement receive RTX stats by Philipp Hancke · 1 year, 11 months ago
  49. 6a7bf10 Replace "rcvd" with "received" for readability by Philipp Hancke · 1 year, 11 months ago
  50. 51c632c Use GlobalSimulatedTimeController in more webrtc video engine unittests by Tommi · 1 year, 11 months ago
  51. ec2670e Cleanup ReportBlockData class: use Timestamp and TimeDelta by Danil Chapovalov · 1 year, 11 months ago
  52. 22f14fe Revert "Create default video factories directly instead of through legacy public helpers" by Danil Chapovalov · 1 year, 11 months ago
  53. 40cb009 Unnest VideoEncoderFactoryTemplate in webrtc_video_engine_unittest.cc by philipel · 1 year, 11 months ago
  54. 44437d3 Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. by philipel · 2 years ago
  55. 3beacb7 Create default video factories directly instead of through legacy public helpers by Danil Chapovalov · 2 years ago
  56. c848268 Use SequenceChecker(SequenceChecker::kDetached) in a few places. by Tommi · 2 years ago
  57. adb9460 Ship ability to opt-in to VP9/AV1 simulcast (re-land). by Henrik Boström · 2 years ago
  58. 80850ca Fix crash happening when changing from legacy to standard VP9. by Henrik Boström · 2 years ago
  59. 75ea06f Revert "Ship ability to opt-in to VP9/AV1 simulcast." by Henrik Boström · 2 years ago
  60. 75990b9 Ship ability to opt-in to VP9/AV1 simulcast. by Henrik Boström · 2 years ago
  61. f6eae95 Delete EncoderSimulcastProxy in favor of SimulcastEncoderAdapter. by Henrik Boström · 2 years ago
  62. 9a5de95 Add a flag to control legacy vs spec-compliant scalability mode. by Henrik Boström · 2 years ago
  63. e744af5 EncoderSimulcastProxy: Respect "supports_simulcast" info. by Henrik Boström · 2 years ago
  64. 2f55370 Reland "Use two MediaChannels for 2 directions." by Harald Alvestrand · 2 years ago
  65. 28c4986 WebRTCVideoChannel::OnPacketReceived: avoid PostTasks. by Markus Handell · 2 years ago
  66. 18c869b Revert "Use two MediaChannels for 2 directions." by Harald Alvestrand · 2 years ago
  67. ba088b1 Revert "Add plumbing for video NACK to be coupled between channels." by Harald Alvestrand · 2 years ago
  68. db1fae4 Reland "Remove ISAC media constant and payload type mapping" by Alessio Bazzica · 2 years ago
  69. bea2278 Separate `last_stats_log_ms_` for send and receive stats. by Linus Nilsson · 2 years ago
  70. b79b74e Revert "Remove ISAC media constant and payload type mapping" by Björn Terelius · 2 years ago
  71. 4c7271a Remove ISAC media constant and payload type mapping by Philipp Hancke · 2 years ago
  72. a087f6f Add plumbing for video NACK to be coupled between channels. by Harald Alvestrand · 2 years ago
  73. 8981a6f Use two MediaChannels for 2 directions. by Harald Alvestrand · 2 years, 1 month ago
  74. 880f1d5 Update simulcast_encoder_adapter_unittest.cc to use absl::optional<>. by Henrik Boström · 2 years, 1 month ago
  75. 2e540a2 Introduce EncodedImage.SimulcastIndex(). by Henrik Boström · 2 years, 1 month ago
  76. 16579cc Change MediaChannel to have a Role parameter by Harald Alvestrand · 2 years, 1 month ago
  77. 101c6aa Remove leftover function signatures. by Fredrik Solenberg · 2 years, 1 month ago
  78. c5455e7 Allow RTX ssrc to be updated on receive streams by Per K · 2 years, 1 month ago
  79. 217b384 Remove rtp header extension from config of Call audio and video receivers by Per K · 2 years, 1 month ago
  80. 664cf14 Reland "Delete PacketReceiver::DeliverPacket from all implementations" by Per K · 2 years, 1 month ago
  81. 7a67dce prefer absl::optional for rtx-time by Philipp Hancke · 2 years, 1 month ago
  82. f2a083f Revert "Delete PacketReceiver::DeliverPacket from all implementations" by Andrey Logvin · 2 years, 1 month ago
  83. 897ea04 Delete PacketReceiver::DeliverPacket from all implementations by Per K · 2 years, 2 months ago
  84. 438b5b4 WebRtcVideoChannel creates default stream with dummy SSRC on received RTX packet. by Per K · 2 years, 1 month ago
  85. 9ad10bc Only generate codec stats for the voice send and receive codec by Philipp Hancke · 2 years, 2 months ago
  86. a0bc404 Remove WebRTC-Dav1dDecoder kill switch. by philipel · 2 years, 2 months ago
  87. 9ece54f Delete unnecssary AudioReceiveStreamInterface::GetRtpExtensions by Per K · 2 years, 2 months ago
  88. 444741e replace use of iterators with for loops or auto by Philipp Hancke · 2 years, 2 months ago
  89. 94b0559 Only fill send/recv stats if there are send/receive streams by Philipp Hancke · 2 years, 2 months ago
  90. efbe753 Add RTCAudioPlayoutStats to GetStats(). by Fredrik Hernqvist · 2 years, 2 months ago
  91. f7e4071 Only generate codec stats for the video send/recv codec in use by Philipp Hancke · 2 years, 2 months ago
  92. e6b4cbe Add SVC fallback. by Åsa Persson · 2 years, 2 months ago
  93. 89ca299 Use parsed packet from RtpTransport::DemuxPacket in engine and call by Per K · 2 years, 2 months ago
  94. 075c20f Implement FakeCall::DeliverRtpPacket and DeliverRtcpPacket by Per K · 2 years, 2 months ago
  95. 17c4ca8 Use RtpPacketReceived in media/engine/webrtc_video_engine_unittest.cc by Per K · 2 years, 2 months ago
  96. 175f06f Reland "Remove 'trackId' dependency in stats selector algorithm." by Henrik Boström · 2 years, 2 months ago
  97. 1251c64 Split stats generation for MediaChannel into sender and receiver APIs by Harald Alvestrand · 2 years, 2 months ago
  98. 9253240 Reland "Remove use of ReceiveStreamRtpConfig:transport_cc" by Per K · 2 years, 2 months ago
  99. b7f9113 Add API for querying codec support. by Åsa Persson · 2 years, 2 months ago
  100. be5c713 Revert "Remove use of ReceiveStreamRtpConfig:transport_cc" by Olga Sharonova · 2 years, 2 months ago