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webrtc / src / webrtc / 68337ca2dc9568763d00a933cba190c0a3bb2dab / media
  1. 3a84a8f Reland of Periodically update codec bit/frame rate settings. by sprang · 8 years ago
  2. fefdcd4 Enable more unittests on iOS, and disable those that fail on simulator by oprypin · 8 years ago
  3. db65e09 Fix the binary size regression on Chromium Windows. by zhihuang · 8 years ago
  4. 0f3c15e Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) by zhihuang · 8 years ago
  5. 2d5cadc Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) by zhihuang · 8 years ago
  6. d901125 Try to fix the binary size increase issue on Chromium. by zhihuang · 8 years ago
  7. 39e693a Implement timing frames. by ilnik · 8 years ago
  8. a50b40f Add cropping to VIEEncoder to match simulcast streams resolution by ilnik · 8 years ago
  9. b49c6a7 Remove unused #include "libyuv/compare.h" by eladalon · 8 years ago
  10. 5e343ec Delete SignalSrtpError. by nisse · 8 years ago
  11. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  12. f94a820 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 8 years ago
  13. 83b56af Use the same QP max for tests as in production by sprang · 8 years ago
  14. f7f8eb4 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 8 years ago
  15. 5dc4393 Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface by Magnus Jedvert · 8 years ago
  16. c419c8d Add field trial for balanced degradation preference. by asapersson · 8 years ago
  17. 1054be8 Remove webrtcvideoengine2.h by eladalon · 8 years ago
  18. 44dd04f Increase number of unsignaled audio streams we handle to 4. by solenberg · 8 years ago
  19. 3d545a2 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
  20. 57314ae Revert of Periodically update codec bit/frame rate settings. (patchset #2 id:160001 of https://codereview.webrtc.org/2924023002/ ) by sprang · 8 years ago
  21. 10b2ceb Reland of Periodically update codec bit/frame rate settings. (patchset #1 id:1 of https://codereview.webrtc.org/2923993002/ ) by sprang · 8 years ago
  22. fac4604 Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ ) by sprang · 8 years ago
  23. bd64bc1 Fix bug in vie_encoder.cc which caused channel parameters not to be updated at regular intervals, as it was intended. by sprang · 8 years ago
  24. 3de35bb Remove outdated warning suppressions. by Kári Tristan Helgason · 8 years ago
  25. 578a329 MediaCodecVideoEncoder: Add QP stats to Encoded callback for VP9 and turn on quality scaling. by asapersson · 8 years ago
  26. 33944ff Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  27. fb8fb2d Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  28. 9f8b6f3 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  29. 4e13fcb Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 8 years ago
  30. df24c99 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  31. 5acd6fa Update I420Buffer to new VideoFrameBuffer interface by magjed · 8 years ago
  32. b6a2c8f Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 8 years ago
  33. ddb82e2 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  34. 1a8221f Recreate FlexfecReceiveStream separately from VideoReceiveStream. by brandtr · 8 years ago
  35. 266f0b3 Avoid toggling default receive streams in WebRtcVideoChannel2. by brandtr · 8 years ago
  36. c4bfd4a Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2910633002/ ) by aleloi · 8 years ago
  37. 1af79fe Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ ) by aleloi · 8 years ago
  38. e47cf2b Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ ) by aleloi · 8 years ago
  39. d46572d Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ ) by aleloi · 8 years ago
  40. 0191502 Activate 'offload debug dump recordings from audio thread to TaskQueue'. by aleloi · 8 years ago
  41. a480e48 Update screen simulcast config and fix periodic encoder param update by sprang · 8 years ago
  42. 7427e1c Field trial support to whenever possible turn off the AGC and HPF by peah · 8 years ago
  43. 816e15e Reduce VideoSendStream recreations due to FlexFEC. by brandtr · 8 years ago
  44. 0d1e81b WebRtcVideoEncoderFactory cleanup by magjed · 8 years ago
  45. d1df7af Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 8 years ago
  46. e6bd325 Update comments for removal of MediaController. by nisse · 8 years ago
  47. 5af64de Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ ) by nisse · 8 years ago
  48. d9704a0 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ ) by nisse · 8 years ago
  49. d477b8c Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ ) by nisse · 8 years ago
  50. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  51. 47f48ce Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ ) by nisse · 8 years ago
  52. 8d238fb Delete RawVideoType enum, use the VideoType enum instead. by nisse · 8 years ago
  53. 0173390 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) by nisse · 8 years ago
  54. 3244692 Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 8 years ago
  55. fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  56. 5c7d0c4 Normalize codec names to those used by AcmCodecDatabase. by ossu · 8 years ago
  57. b84fe88 Allow a received audio codec's payload type to change. by deadbeef · 8 years ago
  58. 0f578cb Use field_trial::IsEnabled for FlexFEC. by brandtr · 8 years ago
  59. 279bdc9 Creating webrtc:video_stream_api by mbonadei · 8 years ago
  60. 4a5e818 Enabling 'gn check' on webrtc/video. by mbonadei · 8 years ago
  61. 30fd8c1 Add content type extension to capabilities by ilnik · 8 years ago
  62. 9268ad2 Enable GN check for webrtc/{p2p,system_wrappers} by kjellander · 8 years ago
  63. 37a786a Fix RtpReceiver.GetParameters when SSRCs aren't signaled. by deadbeef · 8 years ago
  64. 6dc5c17 Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests by deadbeef · 8 years ago
  65. ff493f2 Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 8 years ago
  66. 04c660e Don't add stuff to namespace std by kwiberg · 8 years ago
  67. cb436f6 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ ) by ilnik · 8 years ago
  68. 8f162b3 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) by ilnik · 8 years ago
  69. 1087c1e Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  70. f888bbf Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) by ilnik · 8 years ago
  71. 6b65c51 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  72. fa3ebb0 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) by ilnik · 8 years ago
  73. e849cea Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 8 years ago
  74. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  75. af8f6e5 Move AudioDecoder and AudioDecoderFactory mocks to webrtc/test/ by kwiberg · 8 years ago
  76. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  77. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  78. 138c150 Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  79. 9b1cf9e Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) by ilnik · 8 years ago
  80. 69b425e Revert of Deliver video frames on Android, on the decode thread. (patchset #7 id:120001 of https://codereview.webrtc.org/2764573002/ ) by guidou · 8 years ago
  81. 00ec856 Deliver video frames on Android, on the decode thread. by tommi · 8 years ago
  82. bbca310 Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) by guidou · 8 years ago
  83. 71dd2e5 Move video_encoder.h and video_decoder.h to /api and create GN targets for them by ilnik · 8 years ago
  84. 7950bab Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 8 years ago
  85. 14e0920 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) by lliuu · 8 years ago
  86. acef169 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) by sprang · 8 years ago
  87. a54ea6e Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ ) by nisse · 8 years ago
  88. cd05a9e Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 8 years ago
  89. 90e3b7d Use simulcast for screenshare only in conference mode by sprang · 8 years ago
  90. 31279bc Change minimum DTMF event duration to be 40 milliseconds by dminor · 8 years ago
  91. f67ccf5 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ ) by lliuu · 8 years ago
  92. c8e0342 Remove VoECodec from FakeWebRtcVoiceEngine. by solenberg · 8 years ago
  93. 9b9b7e2 Don't hardcode MediaType::ANY in FakeNetworkPipe. by nisse · 8 years ago
  94. 59773ff Set max bitrate for audio send stream based on RtpParameters. by minyue · 8 years ago
  95. 2fdab80 Vp9: Enable denoiser by default. by jianj · 8 years ago
  96. afd1255 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  97. 5be52fb Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video. by stefan · 8 years ago
  98. 8482abd Adding deadbeef@ as owner of webrtc/media/. by deadbeef · 8 years ago
  99. 52b9df6 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  100. 135be28 WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago