Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 53e048d83aa47de7bf98d1f10c667bd0e79551e7 / audio / audio_state.cc
  1. 649a385 Removes usage of analog AGC. by henrika · 7 years ago
  2. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  3. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  4. aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
  5. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  6. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  7. 6d85252 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up) by henrika · 7 years ago
  8. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  9. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  10. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  11. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  12. 6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
  13. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  14. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  15. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_state.cc]
  16. e67bedb External APM usage downstream dependency support cleanup by peah · 7 years ago
  17. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  18. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  19. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  20. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  21. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  22. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  23. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago