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gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / call / audio_receive_stream.h
  1. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  2. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  3. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  4. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  5. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/audio_receive_stream.h]
  6. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  7. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  8. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  9. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  10. 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
  11. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  12. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  13. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  14. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  15. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  16. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  17. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 7 years ago
  18. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 7 years ago
  19. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 8 years ago
  20. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (96%) from webrtc/api/call/audio_receive_stream.h]
  21. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  22. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  23. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  24. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (96%) from webrtc/audio_receive_stream.h]
  25. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 8 years ago
  26. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 8 years ago
  27. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 8 years ago
  28. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  29. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  30. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 8 years ago
  31. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 8 years ago
  32. 884f585 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  33. 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
  34. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  35. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  36. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  37. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  38. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  39. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  40. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  41. cf18b34 Align new VoE API with design. by solenberg · 9 years ago
  42. 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 9 years ago
  43. cd67022 Define Stream base classes by Jelena Marusic · 9 years ago
  44. 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 9 years ago
  45. 04f4931 VoE2 API draft by Fredrik Solenberg · 9 years ago
  46. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 9 years ago