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gerrit-public.fairphone.software / platform / external / webrtc / e51a0a8bf161c6aaaa28f21924e53465ecc3e02c / audio / channel_proxy.cc
  1. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  2. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago[Renamed (99%) from voice_engine/channel_proxy.cc]
  3. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  4. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  5. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  6. aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
  7. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  8. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  9. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  10. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  11. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  12. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  13. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  14. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/voice_engine/channel_proxy.cc]
  15. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  16. 27e812e Uncomment thread-checkers in ChannelProxy by eladalon · 7 years ago
  17. 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
  18. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  19. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  20. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  21. 0703856 Add SafeClamp(), which accepts args of different types by kwiberg · 8 years ago
  22. edd6eea Rename elad.alon to eladalon, to avoid confusion between repositories. by eladalon · 8 years ago
  23. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  24. cae45d0 Move RtpTransportControllerSend to a new file. by nisse · 8 years ago
  25. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  26. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  27. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  28. 1ffbd6c Injectable audio encoders: voice_engine/channel changes. by ossu · 8 years ago
  29. fdbfdc9 Let PacketRouter separate send and receive modules. by nisse · 8 years ago
  30. 4e76451 Fix UT failure by temporarily uncommenting by elad.alon · 8 years ago
  31. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  32. b8f9a32 Define RtpTransportControllerSendInterface. by nisse · 8 years ago
  33. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  34. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  35. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 8 years ago
  36. d12a8e1 Attach TransportFeedbackPacketLossTracker to ANA (PLR only) by elad.alon · 8 years ago
  37. 8d73f8c Remove VoEVolumeControl interface. by solenberg · 8 years ago
  38. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  39. 0f8b403 Introduce a new constructor to PlatformThread. by tommi · 8 years ago
  40. 0335e6c Fix flaky test WebRtcMediaRecorderTest.PeerConnection by solenberg · 8 years ago
  41. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  42. 08b19df Remove VoEVideoSync interface. by solenberg · 8 years ago
  43. 7de8d64 Wire up audio packet loss to BWE. by stefan · 8 years ago
  44. bd9a77f Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream. by solenberg · 8 years ago
  45. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  46. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  47. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  48. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  49. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  50. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  51. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  52. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  53. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  54. b521aa7 Clean up abs-send-time for audio. by stefan · 8 years ago
  55. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  56. 051f678 Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  57. 6c27849 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  58. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  59. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  60. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  61. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  62. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  63. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 8 years ago
  64. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  65. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  66. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  67. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  68. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  69. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  70. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  71. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  72. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  73. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  74. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  75. 7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  76. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  77. b7f89d6 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ by kwiberg · 9 years ago
  78. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  79. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  80. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  81. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  82. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  83. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  84. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  85. 358057b Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. by solenberg · 9 years ago
  86. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago