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webrtc / src / / 9a34d80fc4a8a312a2448db265dad6aa2bb3f135 / media / BUILD.gn
  1. 9a34d80 Apply the "shim" pattern for WebRtcVoiceEngine by Harald Alvestrand · 1 year, 10 months ago
  2. 97c9623 Make a shim object implementing the VideoMediaChannel interface by Harald Alvestrand · 1 year, 10 months ago
  3. 13897e6 Change SSRC-passing for MediaChannel from external to callback by Harald Alvestrand · 1 year, 10 months ago
  4. 22f14fe Revert "Create default video factories directly instead of through legacy public helpers" by Danil Chapovalov · 2 years ago
  5. 44437d3 Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. by philipel · 2 years ago
  6. 3beacb7 Create default video factories directly instead of through legacy public helpers by Danil Chapovalov · 2 years ago
  7. 1fabbac Update SctpTransportInternal to use RTCError. by Tommi · 2 years ago
  8. f6eae95 Delete EncoderSimulcastProxy in favor of SimulcastEncoderAdapter. by Henrik Boström · 2 years ago
  9. 186ebdc Reland "Remove interim .h files from rtc_media_base" by Harald Alvestrand · 2 years, 1 month ago
  10. 2d57f38 Revert "Remove interim .h files from rtc_media_base" by Harald Alvestrand · 2 years, 1 month ago
  11. 16205b3 Remove interim .h files from rtc_media_base by Harald Alvestrand · 2 years, 1 month ago
  12. a6b9924 Remove all usage of //rtc_base target by Florent Castelli · 2 years, 2 months ago
  13. efbe753 Add RTCAudioPlayoutStats to GetStats(). by Fredrik Hernqvist · 2 years, 2 months ago
  14. e6b4cbe Add SVC fallback. by Åsa Persson · 2 years, 3 months ago
  15. 89ca299 Use parsed packet from RtpTransport::DemuxPacket in engine and call by Per K · 2 years, 3 months ago
  16. 075c20f Implement FakeCall::DeliverRtpPacket and DeliverRtcpPacket by Per K · 2 years, 3 months ago
  17. a138c6c Split rtc_base into multiple targets by Florent Castelli · 2 years, 3 months ago
  18. d3ac3b6 Remove AsSendChannel/AsReceiveChannel methods by Harald Alvestrand · 2 years, 3 months ago
  19. 61dfaaf Delete unused feature to cache packets for unsignaled SSRCs. by Per K · 2 years, 3 months ago
  20. 794d599 Split media_channel and its dependencies from the rtc_media_base target by Harald Alvestrand · 2 years, 3 months ago
  21. 50454ef Apply PIMPL pattern to MediaSender and Receiver objects by Harald Alvestrand · 2 years, 3 months ago
  22. c0d44d9 Split audio and video channels into Send and Receive APIs. by Harald Alvestrand · 2 years, 3 months ago
  23. 9b235cd Add scalability mode to RTCOutboundRtpStreamStats stats by Evan Shrubsole · 2 years, 4 months ago
  24. 514dff8 Fix crash when setting RTCRtpEncodingParameter.maxBitrate to a non-positive value. by Jakob Ivarsson · 2 years, 4 months ago
  25. acabb36 pc: Add asynchronous RtpSender::SetParameters() call by Florent Castelli · 2 years, 4 months ago
  26. b4c96d6 Update IWYU mappings with a few more lines. by Harald Alvestrand · 2 years, 5 months ago
  27. 43f0f29 RtpEncodingParameters::request_resolution patch 4 by Jonas Oreland · 2 years, 6 months ago
  28. 0c4563c Remove the libaom av1 decoder. by philipel · 2 years, 6 months ago
  29. 6c2dae2 Move VideoEncoderConfig from api/ into video/config by Jonas Oreland · 2 years, 6 months ago
  30. 7252348 Create EncoderStreamFactory in VideoStreamEncoder by Jonas Oreland · 2 years, 6 months ago
  31. 1262eb5 Move EncoderStreamFactory into own file by Jonas Oreland · 2 years, 6 months ago
  32. 5cb3a90 Remove sigslot usage from SctpTransportInternal by Fredrik Solenberg · 2 years, 7 months ago
  33. ad4a207 In FakeNetworkInterface drop dependency on rtc::MessageHandler by Danil Chapovalov · 2 years, 7 months ago
  34. 13b9f81 Updated associated payload types without recreating receive streams. by Tommi · 2 years, 7 months ago
  35. c05a1be Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable by Danil Chapovalov · 2 years, 8 months ago
  36. b6ff84b Reland "When VP9 SVC is used, use SvcConfig to set max bitrate for the stream." by Erik Språng · 2 years, 9 months ago
  37. a1a7c63 Let PCF.GetRtpSenderCapabilities return codecs' scalabilityModes. by Byoungchan Lee · 2 years, 9 months ago
  38. 4f1af11 Revert "When VP9 SVC is used, use SvcConfig to set max bitrate for the stream." by Erik Språng · 2 years, 9 months ago
  39. 3afb8e2 When VP9 SVC is used, use SvcConfig to set max bitrate for the stream. by Erik Språng · 2 years, 9 months ago
  40. 7a66900 Delete rtc_base/atomic_ops.h by Niels Möller · 2 years, 9 months ago
  41. c374d11 Move to_queued_task.h and pending_task_safety_flag.h into public API by Artem Titov · 2 years, 9 months ago
  42. 6545516 RtpSenderInterface::SetEncoderSelector by Jonas Oreland · 2 years, 10 months ago
  43. 2a2f3ec Reland "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." by philipel · 2 years, 10 months ago
  44. f8c81ca Revert "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." by Philip Eliasson · 2 years, 10 months ago
  45. 256733c Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate. by philipel · 2 years, 10 months ago
  46. 14d0150 Move VP8 SupportsScalabilityMode utility to its own build target by Niels Möller · 2 years, 11 months ago
  47. 8f04c7c sctp: Handle concurrent data channel reset in transport by Florent Castelli · 2 years, 11 months ago
  48. 3b04813 Update SupportsScalabilityMode functions to use enum ScalabilityMode. by Niels Möller · 2 years, 11 months ago
  49. e3b74f8 sctp: Fix data channel closing sequence by Florent Castelli · 2 years, 11 months ago
  50. 79d566b New enum ScalabilityMode. by Niels Möller · 2 years, 11 months ago
  51. c3e6e3a Remove dependency on rtc_base_approved from most targets by Florent Castelli · 3 years ago
  52. a30aef3 Move event_tracer out of rtc_base_approved by Florent Castelli · 3 years ago
  53. ed4aadc Move copy_on_write_buffer out of rtc_base_approved by Florent Castelli · 3 years ago
  54. a876a92 Move byte_order out of rtc_base_approved by Florent Castelli · 3 years ago
  55. f9c5984 Move buffer out of rtc_base_approved by Florent Castelli · 3 years ago
  56. f4db351 Move race_checker out of rtc_base_approved by Florent Castelli · 3 years ago
  57. 71337f3 Move random out of rtc_base_approved by Florent Castelli · 3 years ago
  58. dd837e2 Remove //rtc_base:timeutils from public deps by Florent Castelli · 3 years ago
  59. 57aa81b Remove //rtc_base:stringutils from public deps by Florent Castelli · 3 years ago
  60. e10a9f6 Remove //rtc_base:safe_conversions from public deps by Florent Castelli · 3 years ago
  61. 33d31fb Remove //rtc_base:rtc_event from public deps by Florent Castelli · 3 years ago
  62. 4467ad7 Remove //rtc_base:macromagic from public deps by Florent Castelli · 3 years ago
  63. 0af55ba Remove //rtc_base:logging from public deps by Florent Castelli · 3 years ago
  64. bddfe02 Remove //rtc_base:atomicops from public deps by Florent Castelli · 3 years ago
  65. f2599a7 Remove usrsctp, dcSCTP is now the unique SCTP implementation by Florent Castelli · 3 years ago
  66. e62c2f2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf by Jonas Oreland · 3 years ago
  67. 8ca0613 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 4/inf by Jonas Oreland · 3 years ago
  68. 4476b82 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 3/inf by Jonas Oreland · 3 years, 1 month ago
  69. b951dc6 Allow specifying delayed task precision of dcsctp::Timer. by Henrik Boström · 3 years, 2 months ago
  70. 9570150 Make libaom_av1_encoder always build the libaom encoder. by philipel · 3 years, 2 months ago
  71. b09d872 Reland "Add dav1d decoder to WebRTC." by philipel · 3 years, 4 months ago
  72. 8498b7e Revert "Add dav1d decoder to WebRTC." by Artem Titov · 3 years, 4 months ago
  73. 1478585 Add dav1d decoder to WebRTC. by philipel · 3 years, 4 months ago
  74. ee03431 WebRtcVideoEngineTest: use simulated time. by Markus Handell · 3 years, 5 months ago
  75. b26863e Reland "Handle scalability mode in QueryCodecSupport" by Johannes Kron · 3 years, 6 months ago
  76. 74281be Revert "Handle scalability mode in QueryCodecSupport" by Andrey Logvin · 3 years, 6 months ago
  77. 715a148 Handle scalability mode in QueryCodecSupport by Johannes Kron · 3 years, 6 months ago
  78. 2562cf0 Reland "Wire up non-sender RTT for audio, and implement related standardized stats." by Ivo Creusen · 3 years, 7 months ago
  79. 2c41cba Revert "Wire up non-sender RTT for audio, and implement related standardized stats." by Björn Terelius · 3 years, 7 months ago
  80. 3cd7a0f Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h by Johannes Kron · 3 years, 7 months ago
  81. 59947d2 SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. by Åsa Persson · 3 years, 7 months ago
  82. fb0dca6 Wire up non-sender RTT for audio, and implement related standardized stats. by Ivo Creusen · 3 years, 7 months ago
  83. 8df32eb dcsctp: Add API to indicate packet send status by Victor Boivie · 3 years, 7 months ago
  84. 99a71f4 Move helpers to parse base rtp packet fields to rtp_rtcp module by Danil Chapovalov · 3 years, 8 months ago
  85. 5e726da dcsctp: Extract logging packet observer as utility by Victor Boivie · 3 years, 9 months ago
  86. 6a11c84 dcsctp: Add DcSctpSocketFactory by Florent Castelli · 3 years, 9 months ago
  87. a334dc6 Make VideoSendStream::UpdateActiveSimulcastLayers not block. by Tommi · 3 years, 10 months ago
  88. 47f5f8c Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket by Danil Chapovalov · 3 years, 10 months ago
  89. 816134a Reland "Fix race between enabled() and set_enabled() in VideoTrack." by Tommi · 3 years, 10 months ago
  90. 096ad02 Revert "Fix race between enabled() and set_enabled() in VideoTrack." by Evan Shrubsole · 3 years, 10 months ago
  91. 5ffefe9 Fix race between enabled() and set_enabled() in VideoTrack. by Tommi · 3 years, 10 months ago
  92. d95b149 datachannel: Merge SendDataParams and DMT types with webrtc equivalent by Florent Castelli · 3 years, 11 months ago
  93. cf2aeff Remove lock from MediaChannel by Tommi · 3 years, 11 months ago
  94. a6983c6 sctp: Add DcsctpTransport based on dcSCTP by Florent Castelli · 3 years, 11 months ago
  95. c3fcee7 Move h264_profile_level_id and vp9_profile to api/video_codecs by Johannes Kron · 4 years ago
  96. 7af57c6 Remove RTP data implementation by Harald Alvestrand · 4 years ago
  97. a80c3e5 sctp: Reorganize build targets by Florent Castelli · 4 years ago
  98. affd219 Delete AsyncInvoker usage from SimulatedPacketTransport by Niels Möller · 4 years ago
  99. 22379fc sctp: Rename SctpTransport to UsrSctpTransport by Florent Castelli · 4 years ago
  100. 6b0f19f sctp: Move SctpTransportFactory to a separate file by Florent Castelli · 4 years ago