Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 34029e209d6d116c338e7d65d6980cc395184658 / call / audio_state.h
  1. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  2. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  3. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 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_state.h]
  6. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  7. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  8. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  9. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  10. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (92%) from webrtc/api/call/audio_state.h]
  11. 16e3caa Removed unused forward declaration. by aleloi · 8 years ago
  12. 81da488 Added audio mixer and removed audio device module in AudioState::Config. by aleloi · 8 years ago
  13. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (92%) from webrtc/audio_state.h]
  14. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  15. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago