1. 5b18967 Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  2. 1f11d1a Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  3. 23aa43d Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  4. 5eb0c3b Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  5. 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  6. 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 7 years ago
  7. 42e4711 Piggybacking simulcast id and ALR experiment id into video content type extension. by ilnik · 7 years ago
  8. adaf22f Support a user-provided string for the TLS ALPN extension. by Diogo Real · 7 years ago
  9. 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  10. 816f591 Remove dead code by kwiberg · 7 years ago
  11. 14e884b Reimplement the builtin audio codec factories using the new stuff in api/ by kwiberg · 7 years ago
  12. 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  13. d69b2c2 Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket. by zstein · 7 years ago
  14. 6a5ac8f Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  15. 7865664 Fix an implicit narrowing conversion found by MSVC by oprypin · 7 years ago
  16. c3ea716 Revert of Reimplement the builtin audio codec factories using the new stuff in api/ (patchset #1 id:60001 of https://codereview.webrtc.org/2997713002/ ) by sakal · 7 years ago
  17. 8a71b0d Reimplement the builtin audio codec factories using the new stuff in api/ by kwiberg · 7 years ago
  18. afb4bf2 Fix an implicit narrowing conversion found by MSVC by kwiberg · 7 years ago
  19. 7bce1a8 iSAC floating-point implementation of the Audio{En,De}coderFactoryTemplate APIs by kwiberg · 7 years ago
  20. f6cd047 iSAC fixed-point implementation of the Audio{En,De}coderFactoryTemplate APIs by kwiberg · 7 years ago
  21. 0df2c66 Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ ) by sprang · 7 years ago
  22. 7647da7 Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings. by asapersson · 7 years ago
  23. 6eb2beb L16 implementation of the Audio{En,De}coderFactoryTemplate APIs by kwiberg · 7 years ago
  24. 786989c Revert of Add a flags field to video timing extension. (patchset #15 id:280001 of https://codereview.webrtc.org/3000753002/ ) by emircan · 7 years ago
  25. d9796d8 Revert of L16 implementation of the Audio{En,De}coderFactoryTemplate APIs (patchset #5 id:80001 of https://codereview.webrtc.org/2995523002/ ) by charujain · 7 years ago
  26. 0e1925c L16 implementation of the Audio{En,De}coderFactoryTemplate APIs by kwiberg · 7 years ago
  27. a58c48e Add a flags field to video timing extension. by sprang · 7 years ago
  28. 109fa97 Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec. by mflodman · 7 years ago
  29. 6ab33cf Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
  30. 0f6b13a Audit of kConstants missing the const qualifier by agrieve · 7 years ago
  31. cf8bdaf Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  32. 9e45ec5 G711 implementation of the Audio{En,De}coderFactoryTemplate APIs by kwiberg · 7 years ago
  33. 3b5bf18 Make RTCStatsReport::ToString() return JSON-parseable string. by ehmaldonado · 7 years ago
  34. d29f8b0 Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 7 years ago
  35. 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  36. 2048bc4 Remove default implementation of PeerConnectionInterface::SetBitrate. by zstein · 8 years ago
  37. fbd3f7b Reinstate "API for periodically regathering ICE candidates" by Steve Anton · 8 years ago
  38. 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
  39. beaccf9 Revert "API for periodically regathering ICE candidates" by Magnus Jedvert · 8 years ago
  40. 13e59ec API for periodically regathering ICE candidates by Steve Anton · 8 years ago
  41. 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  42. 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  43. c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
  44. 7b2b061 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 8 years ago
  45. 90ae41e Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 8 years ago
  46. 4b941f0 Report interframe delay sum in old GetStats by ilnik · 8 years ago
  47. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  48. 37342b9 Report timing frames info in GetStats. by ilnik · 8 years ago
  49. ed578f5 Remove unused static VideoEncoder functions by magjed · 8 years ago
  50. b1498a0 Reland of "VideoFrameBuffer: Remove deprecated functions" by Magnus Jedvert · 8 years ago
  51. 39f7f7a Enable the injection of an APM into a peerconnection by peah · 8 years ago
  52. 50f70b3 Opus implementation of the AudioDecoderFactoryTemplate API by kwiberg · 8 years ago
  53. f23c615 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 8 years ago
  54. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  55. 3765e6c Revert "VideoFrameBuffer: Remove deprecated functions" by Magnus Jedvert · 8 years ago
  56. e819c28 VideoFrameBuffer: Remove deprecated functions by Magnus Jedvert · 8 years ago
  57. 92caf48 Add magjed@ as owner of webrtc/api/video/ by Magnus Jedvert · 8 years ago
  58. 0a9e920 Expose ILBC codec in webrtc/api/audio_codecs/ by solenberg · 8 years ago
  59. 456f1e8 Don't forget to support G722 stereo decoding by kwiberg · 8 years ago
  60. 39e693a Implement timing frames. by ilnik · 8 years ago
  61. e9cc1fe Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ ) by charujain · 8 years ago
  62. ed5b759 Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ ) by charujain · 8 years ago
  63. 871cba3 Opus implementation of the AudioDecoderFactoryTemplate API by kwiberg · 8 years ago
  64. 968e806 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 8 years ago
  65. 4b9460d G722 implementation of the AudioEncoderFactoryTemplate API by kwiberg · 8 years ago
  66. aefaea8 G722 implementation of the AudioDecoderFactoryTemplate API by kwiberg · 8 years ago
  67. 9d89565 Templated AudioDecoderFactory by kwiberg · 8 years ago
  68. 8c18d91 Enable SNI in ssl adapter. by Emad Omara · 8 years ago
  69. 4eab3bd Templated AudioEncoderFactory by kwiberg · 8 years ago
  70. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  71. c224fdb Implement operator<< for AudioCodecInfo and AudioCodecSpec by kwiberg · 8 years ago
  72. 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
  73. fb8fb2d Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  74. 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
  75. 5acd6fa Update I420Buffer to new VideoFrameBuffer interface by magjed · 8 years ago
  76. fd87f87 Add separate base classes for I420 and I444 buffers by magjed · 8 years ago
  77. a4c5957 Renaming probing_interval to bwe_period globally. by minyue · 8 years ago
  78. 8357b2c Remove temporary include of builtin_audio_encoder_factory.h. by ossu · 8 years ago
  79. a24bcce Move webrtc/video_frame to common_video/include. by nisse · 8 years ago
  80. 7abae18 Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ ) by zhihuang · 8 years ago
  81. 8a2a0a0 Revert of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2873803002/ ) by zhihuang · 8 years ago
  82. 71f49e5 Make AudioSinkInterface::Data hold a const pointer to the audio buffer. by yujo · 8 years ago
  83. 03655e2 Add support for I444 in VideoFrameBuffer by magjed · 8 years ago
  84. 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
  85. 6b17ee2 Don't add or rename files in webrtc/ and webrtc/api/ without a proper review by kwiberg · 8 years ago
  86. 1641e82 Add myself as OWNER of webrtc/api/ and webrtc/base/ by kwiberg · 8 years ago
  87. 47e6d13 Create an OWNERS file in webrtc/api/audio_codecs/ by kwiberg · 8 years ago
  88. c0ff88b Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  89. 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
  90. 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
  91. 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
  92. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  93. 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
  94. 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
  95. 3244692 Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 8 years ago
  96. b84fe88 Allow a received audio codec's payload type to change. by deadbeef · 8 years ago
  97. ce4e632 Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) by mbonadei · 8 years ago
  98. 0d80b56 Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) by mbonadei · 8 years ago
  99. b70f74c Creating webrtc/modules:module_api by mbonadei · 8 years ago
  100. 9cdb538 GN: Tighten up test target visibility + refactorings by kjellander · 8 years ago