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gerrit-public.fairphone.software / platform / external / webrtc / fcf79cca7bb92d9ec9b8d2f8146d3c076a6ad365 / media / base / media_channel.h
  1. a3fad93 Revert "Remove the aec_quality_min metric." by Mirko Bonadei · 7 years ago
  2. 99b1bd1 Remove the aec_quality_min metric. by Gustaf Ullberg · 7 years ago
  3. 606a597 Remove adjust_agc_delta from WebRtcVoiceEngine by Steve Anton · 7 years ago
  4. aba85d1 Resolve circular dependency in rtc_media_base. by Patrik Höglund · 7 years ago
  5. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  6. c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
  7. c97cf03 Removes unused sample-rate APIs from the ADM. by henrika · 7 years ago
  8. 5f5918f Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks. by Zhi Huang · 7 years ago
  9. e78bcb9 Enable cpplint in media/ by Steve Anton · 7 years ago
  10. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  11. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  12. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  13. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/base/mediachannel.h]
  14. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  15. 50864a8 Add reporting of googContentType via GetStats on send side by ilnik · 7 years ago
  16. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  17. 2e1b40b Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  18. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  19. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  20. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  21. a79cc28 Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  22. e76bd3a 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 · 7 years ago
  23. f04afde Report interframe delay sum in old GetStats by ilnik · 7 years ago
  24. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  25. 2edc684 Report timing frames info in GetStats. by ilnik · 7 years ago
  26. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  27. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  28. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  29. a5e0df6 Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19 by zstein · 7 years ago
  30. f184138 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 7 years ago
  31. f79ade1 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 · 7 years ago
  32. d72098a 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 · 7 years ago
  33. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 7 years ago
  34. 528b793 Update comments for removal of MediaController. by nisse · 7 years ago
  35. 3bc1510 Fix RtpReceiver.GetParameters when SSRCs aren't signaled. by deadbeef · 7 years ago
  36. 55c5be0 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel. by solenberg · 7 years ago
  37. cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 7 years ago
  38. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 7 years ago
  39. ff0e72f Add QP sum stats for received streams. by sakal · 7 years ago
  40. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
  41. 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 8 years ago
  42. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  43. 293e926 Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  44. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  45. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  46. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  47. 1e23461 Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ ) by deadbeef · 8 years ago
  48. 7a5fa6c Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  49. ebbe4f2 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. by zhihuang · 8 years ago
  50. 80ed35e Implement periodic bandwidth probing in application-limited region. by sergeyu · 8 years ago
  51. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  52. b829d9f Add AudioOption for residual echo detector, and enable the echo detector by default on non-mobile platforms. by ivoc · 8 years ago
  53. a65704b Expose RtpCodecParameters to VideoMediaInfo stats. by hbos · 8 years ago
  54. acd935b Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  55. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  56. 7341ab8 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ ) by nisse · 8 years ago
  57. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  58. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  59. 87da404 Implement qpSum stat for video send ssrc stats. by sakal · 8 years ago
  60. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  61. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago
  62. 74097fd Delete unused file screencastid.h. by nisse · 8 years ago
  63. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
  64. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  65. e33c5d9 Added a level controller initialization value to MediaConstraints. by aleloi · 8 years ago
  66. 0934785 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ ) by nisse · 8 years ago
  67. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  68. 84ef615 Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. by aleloi · 8 years ago
  69. a3333bf This CL adds activation logic of the new APM level control by peah · 8 years ago
  70. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  71. 54f9171 Minor lint-fixes in MediaChannel and VideoEngine2. by terelius · 8 years ago
  72. a1c548b Add RtpHeaderExtension to avoid client breakage by isheriff · 8 years ago
  73. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  74. c9b0c26 Surface the IntelligibilityEnhancer on MediaConstraints by Alejandro Luebs · 8 years ago
  75. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  76. 3fe372d Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 8 years ago
  77. a4ac478 Define rtc::BufferT, like rtc::Buffer but for any trivial type by kwiberg · 8 years ago
  78. 0e533ef Update the call when the network route changes by Honghai Zhang · 8 years ago
  79. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 8 years ago
  80. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 8 years ago
  81. 119760a Don't reconfigure the encoder if the video options aren't changing. by deadbeef · 8 years ago
  82. fcc640f Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 8 years ago
  83. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 8 years ago
  84. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  85. 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 8 years ago
  86. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  87. 5f0b83b Enabling rtcp-rsize negotiation and fixing some issues with it. by Taylor Brandstetter · 8 years ago
  88. 505945a Delete unused VideoCapturer statistics. by Niels Möller · 8 years ago
  89. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  90. 0510331 Drop VideoOptions from VideoSendParameters. by nisse · 8 years ago
  91. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 8 years ago
  92. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 8 years ago
  93. 60653ba New flag is_screencast in VideoOptions. by Niels Möller · 8 years ago
  94. 0db023a Move suspend_below_min_bitrate from VideoOptions to MediaConfig. by nisse · 8 years ago
  95. 686a8ef Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 8 years ago
  96. 65c8fd7 Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag. by solenberg · 8 years ago
  97. 4b4dc86 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 8 years ago
  98. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 8 years ago
  99. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 8 years ago
  100. 1afca73 Change to WebRTC license in webrtc/media by kjellander · 8 years ago