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webrtc / src.git / 4d4f62f6e7f6c55393e2d23386997e59d70d67b8 / media
  1. 4d4f62f VideoSendStreamTest: Add tests for encoder reconfiguration. by Åsa Persson · 3 years, 7 months ago
  2. b9b0890 Fix error adding receive stream after packet received from non-primary SSRC. by Taylor Brandstetter · 3 years, 7 months ago
  3. 885ba07 Reland "red: generate and parse the red fmtp format" by Philipp Hancke · 3 years, 7 months ago
  4. fb50179 Convert WebRtcVoiceEngine tests to rtc::make_ref_counted by Niels Möller · 3 years, 7 months ago
  5. 1f38a38 Add ability to set rtp header extensions without recreating streams. by Tommi · 3 years, 7 months ago
  6. 9af2d92 Replace RTC_DLOG with RTC_LOG in SetRecvParameters by Sergey Silkin · 3 years, 7 months ago
  7. 2562cf0 Reland "Wire up non-sender RTT for audio, and implement related standardized stats." by Ivo Creusen · 3 years, 7 months ago
  8. 767235d media: sctp: Convert frequent logs to RTC_DLOG by Victor Boivie · 3 years, 7 months ago
  9. 2c41cba Revert "Wire up non-sender RTT for audio, and implement related standardized stats." by Björn Terelius · 3 years, 7 months ago
  10. 3cd7a0f Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h by Johannes Kron · 3 years, 7 months ago
  11. 59947d2 SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. by Åsa Persson · 3 years, 7 months ago
  12. fb0dca6 Wire up non-sender RTT for audio, and implement related standardized stats. by Ivo Creusen · 3 years, 7 months ago
  13. 75bbd1f Revert "red: generate and parse the red fmtp format" by Björn Terelius · 3 years, 7 months ago
  14. 9d07309 red: generate and parse the red fmtp format by Philipp Hancke · 3 years, 7 months ago
  15. 062acd9 Move frame drop functionality in VideoAdapter into a separate class. by Åsa Persson · 3 years, 7 months ago
  16. d5a0efe Delete deprecated EncoderSimulcastProxy constructor by Danil Chapovalov · 3 years, 7 months ago
  17. fb19596 Allow setting different number of temporal layers per simulcast layer. by Åsa Persson · 3 years, 7 months ago
  18. 29dddff usrsctp: Remove usage of usrsctp_getladdrs() by Florent Castelli · 3 years, 7 months ago
  19. 773a222 red: enable opus-red by default by Philipp Hancke · 3 years, 7 months ago
  20. 8df32eb dcsctp: Add API to indicate packet send status by Victor Boivie · 3 years, 7 months ago
  21. d08930d Migrate test VideoDecoders to new VideoDecoder::Configure by Danil Chapovalov · 3 years, 8 months ago
  22. cfea218 Use backticks not vertical bars to denote variables in comments by Artem Titov · 3 years, 8 months ago
  23. 35b1cb4 Keep `running_` state in sync with active layers. by Tommi · 3 years, 8 months ago
  24. b54cfde Add optional is_qp_trusted property for EncoderInfo. by Qiu Jianlin · 3 years, 8 months ago
  25. 61bd5d9 Video jitter stats fix: Convert RTP timestamp by Di Wu · 3 years, 8 months ago
  26. 37f664f Use backticks not vertical bars to denote variables in comments for /media by Artem Titov · 3 years, 8 months ago
  27. 99a71f4 Move helpers to parse base rtp packet fields to rtp_rtcp module by Danil Chapovalov · 3 years, 8 months ago
  28. 28a2c63 Adding packetsDiscarded to RTCReceivedRtpStreamStats. by Minyue Li · 3 years, 9 months ago
  29. c98aebb Change how alignment requirements are processed by Ilya Nikolaevskiy · 3 years, 9 months ago
  30. 5e726da dcsctp: Extract logging packet observer as utility by Victor Boivie · 3 years, 9 months ago
  31. fb1a0f0 Cleanup rtp utils in media/base by Danil Chapovalov · 3 years, 9 months ago
  32. e91c992 Implement nack_count metric for outbound audio rtp streams. by Jakob Ivarsson · 3 years, 9 months ago
  33. 00ca004 Unify helpers IsRtpPacket and IsRtcpPacket by Danil Chapovalov · 3 years, 9 months ago
  34. b42ced4 Prepare WebRtcVideoReceiveStream for configuration changes. by Tommi · 3 years, 9 months ago
  35. e54914a Implement nack_count metric for inbound audio rtp streams. by Jakob Ivarsson · 3 years, 9 months ago
  36. 899b29e Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Byoungchan Lee · 3 years, 9 months ago
  37. dcb9ffc DataChannel: Propagate SCTP transport errors to the channels by Florent Castelli · 3 years, 9 months ago
  38. 9e2b315 Minor code cleanup of WebRtcVideoReceiveStream. by Tommi · 3 years, 9 months ago
  39. 6a11c84 dcsctp: Add DcSctpSocketFactory by Florent Castelli · 3 years, 9 months ago
  40. 82f21fd Make WebRtcAudioReceiveStream::stream_ const. by Tommi · 3 years, 9 months ago
  41. 55107c8 Update the sync_group id without recreating audio receive streams. by Tommi · 3 years, 9 months ago
  42. 08be9ba Don't recreate the audio receive stream when updating the local_ssrc. by Tommi · 3 years, 10 months ago
  43. 3008bcd Don't recreate audio receive streams on header extension update. by Tommi · 3 years, 10 months ago
  44. d350006 Add rtp_config() accessor to ReceiveStream. by Tommi · 3 years, 10 months ago
  45. 1c1f540 Factor out common receive stream methods to a common interface. by Tommi · 3 years, 10 months ago
  46. e097282 Avoid recreating the audio stream when a frame decryptor is set. by Tommi · 3 years, 10 months ago
  47. 62ec0f6 Add small cooldown to unsignalled ssrc stream creation. by Henrik Boström · 3 years, 10 months ago
  48. 6eda26c Reland "Remove AudioReceiveStream::Reconfigure() method." by Tommi · 3 years, 10 months ago
  49. 8a18e5b Revert "Remove AudioReceiveStream::Reconfigure() method." by Andrey Logvin · 3 years, 10 months ago
  50. e2561e1 Remove AudioReceiveStream::Reconfigure() method. by Tommi · 3 years, 10 months ago
  51. a334dc6 Make VideoSendStream::UpdateActiveSimulcastLayers not block. by Tommi · 3 years, 10 months ago
  52. 47f5f8c Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket by Danil Chapovalov · 3 years, 10 months ago
  53. 58b8d29 fall back to payload types from lower range after exhausting [96,127] by Philipp Hancke · 3 years, 10 months ago
  54. 4f26a3c red: assign payload type 63 to audio/RED for opus by Philipp Hancke · 3 years, 10 months ago
  55. 236ac50 dcsctp: Add public API for BufferedAmountLow by Victor Boivie · 3 years, 10 months ago
  56. b412efd payload type mapping: restrict lower range to <= 63 by Philipp Hancke · 3 years, 10 months ago
  57. af0dff0 dcsctp: start SCTP_DUMP on a new line by Philipp Hancke · 3 years, 10 months ago
  58. 816134a Reland "Fix race between enabled() and set_enabled() in VideoTrack." by Tommi · 3 years, 10 months ago
  59. 096ad02 Revert "Fix race between enabled() and set_enabled() in VideoTrack." by Evan Shrubsole · 3 years, 10 months ago
  60. 5ffefe9 Fix race between enabled() and set_enabled() in VideoTrack. by Tommi · 3 years, 10 months ago
  61. 70cd086 SEA: Only spawn multi-layered encoders if active layers > 1. by Henrik Boström · 3 years, 10 months ago
  62. 402ceff sctp: Reduce logging level for common calls by Victor Boivie · 3 years, 10 months ago
  63. 12c881b Make RtpHelper<>::sending_ atomic. by Tommi · 3 years, 10 months ago
  64. ea7474e Remove redundant VideoSendStream::rtcp_stats field by Danil Chapovalov · 3 years, 10 months ago
  65. 99261af Revert "Define cricket::MediaType in terms of webrtc::MediaType" by Björn Terelius · 3 years, 11 months ago
  66. 6174173 sctp: dcsctp: Print SCTP packets to logs by Victor Boivie · 3 years, 11 months ago
  67. 3ce6391 Define cricket::MediaType in terms of webrtc::MediaType by Harald Alvestrand · 3 years, 11 months ago
  68. 917dcba Add explicit copy constructors and assign operators for some classes. by Byoungchan Lee · 3 years, 11 months ago
  69. 78d5859 sctp: dcsctp: Remove log statement by Victor Boivie · 3 years, 11 months ago
  70. 56e4825 Add boivie and orphis to media/sctp/OWNERS file by Florent Castelli · 3 years, 11 months ago
  71. 819286e Clean out some leftover junk. by Harald Alvestrand · 3 years, 11 months ago
  72. d95b149 datachannel: Merge SendDataParams and DMT types with webrtc equivalent by Florent Castelli · 3 years, 11 months ago
  73. 8ba623d Fix constness by Tomas Gunnarsson · 3 years, 11 months ago
  74. cf2aeff Remove lock from MediaChannel by Tommi · 3 years, 11 months ago
  75. 5183f00 datachannel: Make SendDataParams reliability fields optional<int> by Florent Castelli · 3 years, 11 months ago
  76. c9625f0 Make MediaChannel classes aware of the network thread. by Tommi · 3 years, 11 months ago
  77. a6983c6 sctp: Add DcsctpTransport based on dcSCTP by Florent Castelli · 3 years, 11 months ago
  78. 02e079f Update video tests to clear network interface pointer consistently. by Tomas Gunnarsson · 3 years, 11 months ago
  79. dab3ce8 Reland "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id." by philipel · 4 years ago
  80. a63bee5 Remove Mutex from BaseChannel. by Tommi · 4 years ago
  81. c1d5891 Replace `new rtc::RefCountedObject` with `rtc::make_ref_counted` in a few files by Tomas Gunnarsson · 4 years ago
  82. 88f4b33 usrsctp: Support sending and receiving empty messages by Florent Castelli · 4 years ago
  83. 48171ec Remove more mentions of RTP datachannels by Harald Alvestrand · 4 years ago
  84. 20ee02c Add codec comparison function to SdpVideoFormat by Johannes Kron · 4 years ago
  85. 63b01e1 Remove ReceiveDataParams::timestamp by Florent Castelli · 4 years ago
  86. c3fcee7 Move h264_profile_level_id and vp9_profile to api/video_codecs by Johannes Kron · 4 years ago
  87. 516e284 Remove DataChannelType and deprecated option disable_sctp_data_channels by Florent Castelli · 4 years ago
  88. eb9c3f2 Handle OnPacketSent on the network thread via MediaChannel. by Tomas Gunnarsson · 4 years ago
  89. e984aa2 Add thread accessors to Call. by Tomas Gunnarsson · 4 years ago
  90. e632402 Remove rtp data channel related code from media_channel.* by Tomas Gunnarsson · 4 years ago
  91. 7af57c6 Remove RTP data implementation by Harald Alvestrand · 4 years ago
  92. 15e078c Fix unsignalled ssrc race in WebRtcVideoChannel. by Henrik Boström · 4 years ago
  93. a80c3e5 sctp: Reorganize build targets by Florent Castelli · 4 years ago
  94. affd219 Delete AsyncInvoker usage from SimulatedPacketTransport by Niels Möller · 4 years ago
  95. 17490b5 Fix regression in UsrSctpReliabilityTest by Niels Möller · 4 years ago
  96. 22379fc sctp: Rename SctpTransport to UsrSctpTransport by Florent Castelli · 4 years ago
  97. 6b0f19f sctp: Move SctpTransportFactory to a separate file by Florent Castelli · 4 years ago
  98. 588f9b3 VideoReceiveStream2: AV1 encoded sink support. by Markus Handell · 4 years ago
  99. 006206d rtx-time implementation by Philipp Hancke · 4 years ago
  100. 0b5ec18 Simplify ChannelManager initialization. by Tomas Gunnarsson · 4 years ago