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gerrit-public.fairphone.software / platform / external / webrtc / dac94538a8f46b3b8696def568f377f7ad814891 / call / audio_receive_stream.h
  1. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 7 years ago
  2. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  3. f120cba Delete AudioMonitor and related code. by Niels Möller · 8 years ago
  4. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 8 years ago
  5. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 8 years ago
  6. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 8 years ago
  7. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 8 years ago
  8. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  9. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  10. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  11. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/call/audio_receive_stream.h]
  12. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 8 years ago
  13. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 8 years ago
  14. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  15. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  16. 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 · 8 years ago
  17. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  18. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  19. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  20. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  21. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  22. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  23. 796b8f9 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. by solenberg · 8 years ago
  24. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  25. d32bf75 Pass SdpAudioFormat through Channel, without converting to CodecInst by kwiberg · 9 years ago
  26. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 9 years ago[Renamed (96%) from webrtc/api/call/audio_receive_stream.h]
  27. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 9 years ago
  28. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 9 years ago
  29. 6348978 Add new decoding statistics for muted output by henrik.lundin · 9 years ago
  30. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago[Renamed (96%) from webrtc/audio_receive_stream.h]
  31. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 9 years ago
  32. 8189b02 Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test. by solenberg · 9 years ago
  33. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  34. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  35. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  36. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  37. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 10 years ago
  38. 884f585 Storing raw audio sink for default audio track. by deadbeef · 10 years ago
  39. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 10 years ago
  40. e591f93 Storing raw audio sink for default audio track. by deadbeef · 10 years ago
  41. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 10 years ago
  42. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 10 years ago
  43. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 10 years ago
  44. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 10 years ago
  45. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 10 years ago
  46. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 10 years ago
  47. cf18b34 Align new VoE API with design. by solenberg · 10 years ago
  48. 6bb1b6e Control combined_audio_video_bwe with config bool. by pbos · 10 years ago
  49. cd67022 Define Stream base classes by Jelena Marusic · 10 years ago
  50. 8fc7fa7 Base A/V synchronization on sync_labels. by pbos · 10 years ago
  51. 04f4931 VoE2 API draft by Fredrik Solenberg · 10 years ago
  52. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago