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gerrit-public.fairphone.software / platform / external / webrtc / 0e96535be97916d8fcaa9873ffab3c636539f9d8 / call / audio_state.h
  1. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  2. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  3. 921d366 Remove comments about using std::shared_ptr. by Mirko Bonadei · 6 years ago
  4. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  5. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  6. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 6 years ago
  7. f120cba Delete AudioMonitor and related code. by Niels Möller · 6 years ago
  8. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  9. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  10. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  11. cf73c96 Add AudioDeviceModule to AudioState::Config. by Fredrik Solenberg · 7 years ago
  12. 63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 7 years ago
  13. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  14. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  15. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  16. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  17. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/audio_state.h]
  18. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  19. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  20. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  21. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  22. 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]
  23. 16e3caa Removed unused forward declaration. by aleloi · 8 years ago
  24. 81da488 Added audio mixer and removed audio device module in AudioState::Config. by aleloi · 8 years ago
  25. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (92%) from webrtc/audio_state.h]
  26. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  27. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago