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gerrit-public.fairphone.software / platform / external / webrtc / 53e048d83aa47de7bf98d1f10c667bd0e79551e7 / audio / audio_state_unittest.cc
  1. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  2. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  3. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  4. fc3a2e3 Remove the VoiceEngineObserver callback interface. by 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_state_unittest.cc]
  7. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  8. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 8 years ago
  9. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  10. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  11. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  12. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  13. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 8 years ago
  14. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by 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