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webrtc / src / webrtc / 054582ba6b811fcf4b35601316ff00de29d3e528 / media
  1. 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 8 years ago
  2. f5fb933 WebRtcVideoEngine: Encapsulate logic for unifying internal and external video codecs by magjed · 8 years ago
  3. 29da79c Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder by magjed · 8 years ago
  4. d1f9dfc Clean up ownership of webrtc::VideoEncoder by magjed · 8 years ago
  5. 5bf39fc Removing dependencies on stub headers within WebRTC. by mbonadei · 8 years ago
  6. 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  7. 3ef80a2 Reverse |rtx_payload_types| map, and rename. by nisse · 8 years ago
  8. 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  9. b991f6d Increase logging severity level for SW fallback. by brandtr · 8 years ago
  10. ed56bdc Make CodecType conversion functions non-optional. by kthelgason · 8 years ago
  11. 781a796 Reland of Modify profiles for H264 encode SW fallback (patchset #1 id:1 of https://codereview.webrtc.org/2995373002/ ) by emircan · 8 years ago
  12. 6a5ac8f Report max interframe delay over window insdead of interframe delay sum by ilnik · 8 years ago
  13. 9f96c2a Add video timing frames to set of default RTP header extensions by sprang · 8 years ago
  14. f0c86c0 Move video send/receive stream headers to webrtc/call. by aleloi · 8 years ago
  15. 5630733 Revert of Modify profiles for H264 encode SW fallback (patchset #2 id:20001 of https://codereview.webrtc.org/2997913003/ ) by zhihuang · 8 years ago
  16. ef557d4 Modify profiles for H264 encode SW fallback by emircan · 8 years ago
  17. 7647da7 Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings. by asapersson · 8 years ago
  18. 0a27bbc Delete unneeded Start and Stop methods on FlexfecReceiveStream. by Niels Möller · 8 years ago
  19. 675930d Add support for a forced software encoder fallback. by asapersson · 8 years ago
  20. de6c74c Replace absolute path with relative path for GN files. by Jianjun Zhu · 8 years ago
  21. 4cd205d Default enable content type rtp header extension by sprang · 8 years ago
  22. e55e1f2 Removed unused async_invoker_ in WebRtcVideoCapturer by srte · 8 years ago
  23. 6ab33cf Wire up RTP keep-alive in ortc api. by sprang · 8 years ago
  24. 0f6b13a Audit of kConstants missing the const qualifier by agrieve · 8 years ago
  25. 63b0b01 Renamed fields in common_types.h/RtcpStatistics. by srte · 8 years ago
  26. 783ce68 Rename ViEEncoder to VideoStreamEncoder by mflodman · 8 years ago
  27. 2c1cbc1 Protected streams report RTP messages directly to the FlexFec streams by eladalon · 8 years ago
  28. ea04cc0 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 8 years ago
  29. a6137d6 Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine by peah · 8 years ago
  30. 0dedc27 Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream by eladalon · 8 years ago
  31. 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  32. 5349550 Revert "Prefer external video codecs over internal in SDP" by minyue · 8 years ago
  33. d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 8 years ago
  34. a4a58a0 Prefer external video codecs over internal in SDP by magjed · 8 years ago
  35. 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  36. 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  37. c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
  38. a5b6c52 Remove webrtc::VideoEncoderFactory by magjed · 8 years ago
  39. b7b8932 External APM usage downstream dependency support cleanup by peah · 8 years ago
  40. 4b941f0 Report interframe delay sum in old GetStats by ilnik · 8 years ago
  41. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  42. cda51d6 Update screen simulcast config by sprang · 8 years ago
  43. 37342b9 Report timing frames info in GetStats. by ilnik · 8 years ago
  44. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  45. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  46. 56f7815 Enable -Wunused-function warning everywhere. by Henrik Kjellander · 8 years ago
  47. 0d58090 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  48. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  49. 3a84a8f Reland of Periodically update codec bit/frame rate settings. by sprang · 8 years ago
  50. fefdcd4 Enable more unittests on iOS, and disable those that fail on simulator by oprypin · 8 years ago
  51. db65e09 Fix the binary size regression on Chromium Windows. by zhihuang · 8 years ago
  52. 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
  53. 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
  54. d901125 Try to fix the binary size increase issue on Chromium. by zhihuang · 8 years ago
  55. 39e693a Implement timing frames. by ilnik · 8 years ago
  56. a50b40f Add cropping to VIEEncoder to match simulcast streams resolution by ilnik · 8 years ago
  57. b49c6a7 Remove unused #include "libyuv/compare.h" by eladalon · 8 years ago
  58. 5e343ec Delete SignalSrtpError. by nisse · 8 years ago
  59. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  60. f94a820 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 8 years ago
  61. 83b56af Use the same QP max for tests as in production by sprang · 8 years ago
  62. f7f8eb4 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 8 years ago
  63. 5dc4393 Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface by Magnus Jedvert · 8 years ago
  64. c419c8d Add field trial for balanced degradation preference. by asapersson · 8 years ago
  65. 1054be8 Remove webrtcvideoengine2.h by eladalon · 8 years ago
  66. 44dd04f Increase number of unsignaled audio streams we handle to 4. by solenberg · 8 years ago
  67. 3d545a2 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
  68. 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
  69. 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
  70. 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
  71. 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
  72. 3de35bb Remove outdated warning suppressions. by Kári Tristan Helgason · 8 years ago
  73. 578a329 MediaCodecVideoEncoder: Add QP stats to Encoded callback for VP9 and turn on quality scaling. by asapersson · 8 years ago
  74. 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
  75. fb8fb2d Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  76. 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
  77. 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
  78. df24c99 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  79. 5acd6fa Update I420Buffer to new VideoFrameBuffer interface by magjed · 8 years ago
  80. 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
  81. 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
  82. 1a8221f Recreate FlexfecReceiveStream separately from VideoReceiveStream. by brandtr · 8 years ago
  83. 266f0b3 Avoid toggling default receive streams in WebRtcVideoChannel2. by brandtr · 8 years ago
  84. 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
  85. 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
  86. 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
  87. 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
  88. 0191502 Activate 'offload debug dump recordings from audio thread to TaskQueue'. by aleloi · 8 years ago
  89. a480e48 Update screen simulcast config and fix periodic encoder param update by sprang · 8 years ago
  90. 7427e1c Field trial support to whenever possible turn off the AGC and HPF by peah · 8 years ago
  91. 816e15e Reduce VideoSendStream recreations due to FlexFEC. by brandtr · 8 years ago
  92. 0d1e81b WebRtcVideoEncoderFactory cleanup by magjed · 8 years ago
  93. 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
  94. e6bd325 Update comments for removal of MediaController. by nisse · 8 years ago
  95. 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
  96. 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
  97. 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
  98. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  99. 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
  100. 8d238fb Delete RawVideoType enum, use the VideoType enum instead. by nisse · 8 years ago