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gerrit-public.fairphone.software / platform / external / webrtc / 5154c5bc01ca3d6d7f380e30fa9c9c97917b4dee / media
  1. cf26705 Field trial to control inter-layer prediction. by Sergey Silkin · 6 years ago
  2. b55015e Replacing SequencedTaskChecker with SequenceChecker. by Sebastian Jansson · 6 years ago
  3. 8581877 Delete interface class VideoCaptureExternal by Niels Möller · 6 years ago
  4. f71362f Wire up RTCOutboundRtpStreamStats.totalEncodeTime. by Henrik Boström · 6 years ago
  5. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  6. 5a0665b Make UDP receive buffer size configurable via field trial by Johannes Kron · 6 years ago
  7. a556448 Don't recreate the VideoReceiveStream on SetFrameDecryptor in the MediaEngine. by Benjamin Wright · 6 years ago
  8. 33d2a91 Fix target bitrate RTCP messages behavior for SVC streams by Ilya Nikolaevskiy · 6 years ago
  9. 66e7679 Export symbols needed by the Chromium component build (part 8). by Mirko Bonadei · 6 years ago
  10. 1c41be6 Propagate TaskQueueFactory to AudioDeviceBuffer by Danil Chapovalov · 6 years ago
  11. 9d8eaac Delete unneeded direct includes of common_types.h by Niels Möller · 6 years ago
  12. a7d8f19 Use 'public_deps' to depend on nacl_io [1]. by Mirko Bonadei · 6 years ago
  13. e896490 Revert "Fix target bitrate RTCP messages behavior for SVC streams" by Oleh Prypin · 6 years ago
  14. b118d42 Use Abseil container algorithms in a couple places in media/ by Steve Anton · 6 years ago
  15. 4c7112a Reland "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Danil Chapovalov · 6 years ago
  16. ab65d8a Fix target bitrate RTCP messages behavior for SVC streams by Ilya Nikolaevskiy · 6 years ago
  17. f0d1c03 Add replacement interface for webrtc::GainConrol by Sam Zackrisson · 6 years ago
  18. e27ccf9 Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory" by Amit Hilbuch · 6 years ago
  19. a39254d in WebrtcVoiceEngine allow to set TaskQueueFactory by Danil Chapovalov · 6 years ago
  20. b947fe1 Paying off some debt from rtp_utils.cc by Amit Hilbuch · 6 years ago
  21. 54659c1 Fix obsolete settings in VideoEngine for VP9 screenshare by Ilya Nikolaevskiy · 6 years ago
  22. 8f7ce22 Make VideoFrameType an enum class, and move to separate file and target by Niels Möller · 6 years ago
  23. 94b57c0 Cleanup BUILD.gn files from imports like foo:foo by Artem Titov · 6 years ago
  24. d26a916 Avoid using GlobalTaskQueueFactory for TaskQueueForTest by Danil Chapovalov · 6 years ago
  25. edd2054 Minor fixes and refactoring for RtpTransport until the Demux. by Amit Hilbuch · 6 years ago
  26. 7fbfaa4 PeerConnection::SetBitrate now also configures media transport. by Piotr (Peter) Slatala · 6 years ago
  27. 946b968 Add support for target rate constraints by Piotr (Peter) Slatala · 6 years ago
  28. 6d83592 Improve handling of packets with unknown ssrc. by Jonas Oreland · 6 years ago
  29. 647d5e6 Increase the default maximum jitter buffer size to 200 packets. by Jakob Ivarsson · 6 years ago
  30. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
  31. e25f595 Guard preferred_dscp with the network interface lock by Steve Anton · 6 years ago
  32. 2c7b982 Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode" by Jeroen de Borst · 6 years ago
  33. 39d3a7d Delete CodecSpecificInfo argument from VideoDecoder::Decode by Niels Möller · 6 years ago
  34. b859b32 Update more VideoEncoder implementations to drop CodecSpecificInfo input by Niels Möller · 6 years ago
  35. 87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
  36. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 6 years ago
  37. c8d2e73 Delete CodecSpecificInfo argument from VideoEncoder::Encode by Niels Möller · 6 years ago
  38. ef50b25 Remove lock in WebRtcVideoEngine by Steve Anton · 6 years ago
  39. d1d0359 Remove memsets of CodecSpecificInfo. by philipel · 6 years ago
  40. fc52b91 Implicitly suppress //build/config/clang:find_bad_constructs. by Mirko Bonadei · 6 years ago
  41. c85328f Add SCTP transport to the public API. by Harald Alvestrand · 6 years ago
  42. 493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 6 years ago
  43. b000b71 Wiring up RIDs from the video engine to the RTP Sender. by Amit Hilbuch · 6 years ago
  44. 22f9925 webrtc: Remove semicolons. by Nico Weber · 6 years ago
  45. ccb9b75 Create version 01 of Generic Frame Descriptor - with discardability flag by Elad Alon · 6 years ago
  46. ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 6 years ago
  47. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
  48. 71aee3a Reland "Propagate VideoFrame::UpdateRect to encoder" by Ilya Nikolaevskiy · 6 years ago
  49. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 6 years ago
  50. c664314 Clean up implementation in stream_params by Steve Anton · 6 years ago
  51. 429b67d Revert "Propagate VideoFrame::UpdateRect to encoder" by Mirko Bonadei · 6 years ago
  52. efa72a1 Propagate VideoFrame::UpdateRect to encoder by Ilya Nikolaevskiy · 6 years ago
  53. 3a656d1 Tune bitrates and minQP thresholds for high-fps screenshare. by Ilya Nikolaevskiy · 6 years ago
  54. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  55. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  56. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  57. 6c02541 Revert "Delete video source proxying in WebRtcVideoSendStream" by Christian Fremerey · 6 years ago
  58. 7f24fb9 Add settings to turn off VP8 base layer qp limit by Erik Språng · 6 years ago
  59. b66003c Delete video source proxying in WebRtcVideoSendStream by Niels Möller · 6 years ago
  60. fd5d473 Revert "Partial frame capture API part 6" by Ilya Nikolaevskiy · 6 years ago
  61. 1a1c52b H.264 temporal layers w/frame marking (PART 2/3) by Johnny Lee · 6 years ago
  62. 157540a Stop hard-coding default IDs for RTP extensions by Elad Alon · 6 years ago
  63. 9f3aabb Delete obsolete class cricket::VideoCapturer by Niels Möller · 6 years ago
  64. b76b9ba Set WEBRTC_USE_H264 in common_config by Johannes Kron · 6 years ago
  65. ef288dd Reland: Remove dead code from stream_params.h by Steve Anton · 6 years ago
  66. e1dcce2 Remove HAVE_WEBRTC_VOICE. by Fredrik Solenberg · 6 years ago
  67. cf416e4 Revert "Remove dead code from stream_params.h" by Oleh Prypin · 6 years ago
  68. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  69. 38c83b9 Remove unused file. by Fredrik Solenberg · 6 years ago
  70. 3f408d0 Remove dead code from stream_params.h by Steve Anton · 6 years ago
  71. 7752ad6 Partial frame capture API part 6 by Ilya Nikolaevskiy · 6 years ago
  72. 9387b52 Apply simulcast resolution normalization before scaling. by Rasmus Brandt · 6 years ago
  73. 9f6a0d5 In VideoEngine also respect requested TL number even for screenshare by Ilya Nikolaevskiy · 6 years ago
  74. 167316b Remove proxy layer from AndroidVideoTrackSource by Magnus Jedvert · 6 years ago
  75. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 6 years ago
  76. 65cc52e Fix heap use overrun in FakeEncoder By removing unnecessary fixed size buffer. by Per Kjellander · 6 years ago
  77. 1e6e773 Add steveanton@ to media/ OWNERS by Steve Anton · 6 years ago
  78. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
  79. 042bb00 Fix RTP transport accepting invalid RTCP headers. by Piotr (Peter) Slatala · 6 years ago
  80. 6957abe Reland "Always use real VideoStreamsFactory in full stack tests" by Ilya Nikolaevskiy · 6 years ago
  81. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 6 years ago
  82. 2c9ebef Use Abseil container algorithms in media/ by Steve Anton · 6 years ago
  83. bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 6 years ago
  84. 190713c Remove +api from internal DEPS files. by Mirko Bonadei · 6 years ago
  85. 739baf0 [clang-tidy] Apply performance-for-range-copy fixes. by Mirko Bonadei · 6 years ago
  86. f380284 (7) Rename files to snake_case: remove forwarding headers by Steve Anton · 6 years ago
  87. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  88. 1e27fec Negate flag name for prerender smoothing and update comments. by Rasmus Brandt · 6 years ago
  89. 470a5ea Introduces common AudioAllocationSettings class. by Sebastian Jansson · 6 years ago
  90. b0397d6 Always send abs-send-time when negotiated and do not filter it out. by Konrad Hofbauer · 6 years ago
  91. 1b761ca Remove simulcast constraints in SimulcastEncoderAdapter by Florent Castelli · 6 years ago
  92. 805a27e Reland "Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2." by Niels Möller · 6 years ago
  93. a882fb3 Revert "Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2." by Niels Moller · 6 years ago
  94. aa8b94c Refactor WebRtcVideoEngine tests to not use cricket::VideoCapturer, part 2. by Niels Möller · 6 years ago
  95. 3de32e6 Add test WebRtcVideoChannelTest.DoesNotAdaptWhenScreeensharing by Niels Möller · 6 years ago
  96. dbdd839 Add ability for VideoEncoder to signal frame rate allocation. by Erik Språng · 6 years ago
  97. f93eda1 Move some video codec constants to separate file. by Erik Språng · 6 years ago
  98. fb7a6a7 Delete test WebRtcVideoChannelTest.AdaptsOnOveruseAndChangeResolution by Niels Möller · 6 years ago
  99. 1ba3ddd Remove obsolete video_capture_module dependency. by Yves Gerey · 6 years ago
  100. 7c03bdc Reland "Add a high bitrate full stack test with fake codec" by Emircan Uysaler · 6 years ago