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gerrit-public.fairphone.software / platform / external / webrtc / 53e048d83aa47de7bf98d1f10c667bd0e79551e7 / audio / audio_receive_stream.h
  1. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  2. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  3. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  4. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_receive_stream.h]
  7. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  8. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  9. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  10. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  11. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  12. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 7 years ago
  13. e4bcd6d New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 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. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 7 years ago
  19. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 7 years ago
  20. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 7 years ago
  21. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 7 years ago
  22. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  23. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  24. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  25. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  26. 051f678 Add a NeededFrequency() method to the AudioMixer::Source interface. by aleloi · 8 years ago
  27. 6c27849 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  28. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  29. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  30. 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
  31. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 8 years ago
  32. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  33. 217fb66 Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume(). by solenberg · 8 years ago
  34. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  35. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 8 years ago
  36. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 8 years ago
  37. 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
  38. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  39. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  40. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  41. 1372508 Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. by solenberg · 9 years ago
  42. 7add058 Move some receive stream configuration into webrtc::AudioReceiveStream. by solenberg · 9 years ago
  43. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago
  44. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  45. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  46. 4f4ec0a Re-Land: Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  47. 43e83d4 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) by solenberg · 9 years ago
  48. a457752 Implement AudioReceiveStream::GetStats(). by Fredrik Solenberg · 9 years ago
  49. a2f30de Log Call {audio, video} stream deletions. by pbos · 9 years ago
  50. 5c389d3 Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 9 years ago[Renamed (91%) from webrtc/video/audio_receive_stream.h]
  51. 68786d2 Wire up PacketTime to ReceiveStreams. by stefan · 9 years ago
  52. cd67022 Define Stream base classes by Jelena Marusic · 9 years ago
  53. 04f4931 VoE2 API draft by Fredrik Solenberg · 9 years ago
  54. 23fba1f Add AudioReceiveStream to Call API. by Fredrik Solenberg · 9 years ago