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gerrit-public.fairphone.software / platform / external / webrtc / 665174fdbb4e0540eccb27cf7412348f1b65534c / audio / audio_state.h
  1. 665174f Reformat the WebRTC code base by Yves Gerey · 6 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. 63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 7 years ago
  7. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  8. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  9. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  10. 9155e49 New classes RefCounter and RefCountedBase. by Niels Möller · 7 years ago
  11. 6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
  12. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  13. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  14. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_state.h]
  15. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  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. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  22. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  23. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  24. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  25. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago