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gerrit-public.fairphone.software / platform / external / webrtc / 894eb8b0c9ba8aa304eca33465dad3754afdc7e5 / media
  1. 09f1195 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 5 years ago
  2. 27b0e0d Remove obsolete todo comment in simulcast.h by Åsa Persson · 5 years ago
  3. e942b14 New build target api:media_interface by Niels Möller · 5 years ago
  4. 1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
  5. 53227cc Remove webrtc::MinPositive from api/. by Mirko Bonadei · 5 years ago
  6. 738bfa7 Remove api/bitrate_constraints.h. by Mirko Bonadei · 5 years ago
  7. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  8. d9cc8c0 Encoder switching based on network and/or resolution conditions. by philipel · 5 years ago
  9. 73ceed5 Update simulcast bitrate calculations for non-standard resolutions. by Ilya Nikolaevskiy · 5 years ago
  10. 7bf7a42 Delete flag VideoReceiveStream::Config::Rtp::remb by Niels Möller · 5 years ago
  11. eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
  12. 70dd165 Delete CoreAudio include from media_engine.h by Niels Möller · 5 years ago
  13. 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
  14. fcfeefe Move rtc_error.{h,cc} to its own build target. by Mirko Bonadei · 5 years ago
  15. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
  16. 0bd2eff Reland "New build target p2p:stun_types" by Niels Möller · 5 years ago
  17. 91c824f Revert "New build target p2p:stun_types" by Hannes Landeholm · 5 years ago
  18. 66d6c3b Buffers non atomic message send with usrsctp lib. by Seth Hampson · 5 years ago
  19. 8c5520c Reland "Make the min video bitrate in VideoSendStream configurable." by Ying Wang · 5 years ago
  20. 1d2149c Revert "Make the min video bitrate in VideoSendStream configurable." by Alessio Bazzica · 5 years ago
  21. b2fb0b9 Make the min video bitrate in VideoSendStream configurable. by Ying Wang · 5 years ago
  22. a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
  23. 5b4fcb5 New build target p2p:stun_types by Niels Möller · 5 years ago
  24. 25eb47c Make the RtpHeaderParserImpl available to tests and tools only. by Tommi · 5 years ago
  25. b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
  26. 6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
  27. 4271afb Fix the bug and reland "Make min video target bitrate configurable." by Ying Wang · 5 years ago
  28. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  29. 7e896d0 Revert "Make min video target bitrate configurable." by Mirko Bonadei · 5 years ago
  30. a471e79 Make min video target bitrate configurable. by Ying Wang · 5 years ago
  31. d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
  32. 224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
  33. b689af4 Changes to enable use of DatagramTransport as a data channel transport. by Bjorn A Mellem · 5 years ago
  34. 70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
  35. 587991c Remove jeroendb@webrtc.org from OWNERS by Steve Anton · 5 years ago
  36. 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 5 years ago
  37. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 5 years ago
  38. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 5 years ago
  39. bbeb109 Reporting audio device underrun counter by Alex Narest · 5 years ago
  40. 71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
  41. 78a7138 Remove MediaTransport from Call. by Tommi · 5 years ago
  42. 5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
  43. 2d2bbb1 Filter out duplicate receive codecs in the media engine by Steve Anton · 5 years ago
  44. 83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
  45. f40a340 Remove deprecated code related to AEC2 by Per Åhgren · 5 years ago
  46. d2845f8 Removes unused AudioAllocationSettings from voice engine. by Sebastian Jansson · 5 years ago
  47. 9b1700c Enable field trial LegacySimulcastLayerLimit by default by Florent Castelli · 5 years ago
  48. d7ee76c Wire up field trials for some experimental screenshare settings by Erik Språng · 5 years ago
  49. 8bbdb5b Update VideoBitrateAllocator allocate to take a struct with more fields by Florent Castelli · 5 years ago
  50. da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 5 years ago
  51. bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
  52. bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 5 years ago
  53. 0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
  54. a9fbb22 Add a field trial for older applications to reduce the simulcast layer count by Florent Castelli · 5 years ago
  55. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 5 years ago
  56. 74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 5 years ago
  57. 0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
  58. e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
  59. 0bb0881 Add VideoEncoderFactory::GetImplementations function. by philipel · 5 years ago
  60. 66b3860 Remove WebRTC-SimulcastScreenshare and enable it by default by Florent Castelli · 5 years ago
  61. 41300af Poison default task queue factory by Danil Chapovalov · 5 years ago
  62. 80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
  63. 495a1ae Remove cricket::WebRtcMediaEngineFactory as now unused by Danil Chapovalov · 5 years ago
  64. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  65. 668ce0c Remove trial WebRTC-SimulcastMaxLayers and make its behavior default by Florent Castelli · 5 years ago
  66. fdf74bd Remove non implemented function from WebRtcVideoChannel. by philipel · 5 years ago
  67. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  68. 53d45ba Make TaskQueueFactory required construction parameter for Call by Danil Chapovalov · 5 years ago
  69. e8ed830 WebRtcVideoChannel encoder fallback. by philipel · 5 years ago
  70. 5ee6967 Don't reset encoder on max/min bitrate change. by Sergey Silkin · 5 years ago
  71. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  72. 5983087 Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8. by Åsa Persson · 5 years ago
  73. 65764e4 Add missing overrides in VideoEncoder proxies/adapters by Elad Alon · 5 years ago
  74. 8f01c4e Define FecControllerOverride and plumb it down to VideoEncoder by Elad Alon · 5 years ago
  75. 2efae77 Add RTCStats for keyFramesEncoded, keyFramesDecoded. by Rasmus Brandt · 5 years ago
  76. 4ba04b7 Delete RtcEventLogFactory factory as now unused by Danil Chapovalov · 5 years ago
  77. 90f3b89 Replace the implementation of `GetContributingSources()` on the video side. by Chen Xing · 5 years ago
  78. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  79. c538506 Enable H.264 temporal scalability in simulcast. by Johnny Lee · 5 years ago
  80. f00bf42 Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker. by Chen Xing · 5 years ago
  81. 98cbb22 Moved AsyncInvoker to be destructed first in WebRtcVideoSendStream. by philipel · 5 years ago
  82. 6e9c2fd Delete StartRtcEventLog and StopRtcEventLog methods from FakeVoiceEngine by Niels Möller · 5 years ago
  83. a9952cb Uncomment "override" in simulcast_encoder_adapter_unittest.cc by Elad Alon · 5 years ago
  84. 370f93a Reland "Inform VideoEncoder of negotiated capabilities" by Elad Alon · 5 years ago
  85. e8e4dc4 Change StartAecDump methods to work with FILE* and FileWrapper by Niels Möller · 5 years ago
  86. 49d661a Revert "Inform VideoEncoder of negotiated capabilities" by Philip Eliasson · 5 years ago
  87. 11dfff0 Inform VideoEncoder of negotiated capabilities by Elad Alon · 5 years ago
  88. 5d24b16 Prepare for splitting the api/video:video_frames build rule. by Chen Xing · 5 years ago
  89. 27a7e9f Remove myself from OWNERS in a few places. by Fredrik Solenberg · 5 years ago
  90. 71061bc Replace calls to deprecated googletest APIs. by Mirko Bonadei · 5 years ago
  91. 4fc0855 Cleanup video frame metadata copying by Ilya Nikolaevskiy · 5 years ago
  92. 479a3c0 Add support for enabling and negotiating raw RTP packetization. by Mirta Dvornicic · 5 years ago
  93. 2dbc627 Check H264 packetization mode when using IsSameCodec by Steve Anton · 5 years ago
  94. 85b8ce2 In media engine replace forward declaration with proper includes by Danil Chapovalov · 5 years ago
  95. 220f4be Remove some media/ --> pc/ test dependencies by Steve Anton · 5 years ago
  96. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  97. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  98. 6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  99. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  100. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago