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gerrit.openfyde.cn / webrtc.googlesource.com / src / 603171647318927390e10c04d918d4ed7075717e / audio / audio_state_unittest.cc
  1. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  2. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  3. 6a489f2 Fully qualify googletest symbols. 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 · 7 years ago
  5. ac63ac7 Update refcounting of AudioState to use rtc::RefCountedObject by Niels Möller · 7 years ago
  6. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  7. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  8. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 8 years ago
  9. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 8 years ago
  10. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 8 years ago
  11. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 8 years ago
  12. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  13. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/audio/audio_state_unittest.cc]
  14. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  15. 04c0722 Replace AudioConferenceMixer with AudioMixer. by aleloi · 9 years ago
  16. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 9 years ago
  17. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 9 years ago
  18. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 9 years ago
  19. 77eab70 Enable the -Wundef warning for clang by kwiberg · 9 years ago
  20. fffa42b Replace scoped_ptr with unique_ptr in webrtc/audio/ by kwiberg · 9 years ago
  21. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 10 years ago
  22. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 10 years ago