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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
e36a7cbd1d060961b07c0b9a1e869f777cbbdcb5
/
audio
/
audio_state_unittest.cc
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago
d319534
Move ADM initialization into WebRtcVoiceEngine
by Fredrik Solenberg
· 7 years ago
fc3a2e3
Remove the VoiceEngineObserver callback interface.
by solenberg
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/audio/audio_state_unittest.cc]
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 7 years ago
04c0722
Replace AudioConferenceMixer with AudioMixer.
by aleloi
· 8 years ago
10111bc
Passed AudioMixer to AudioState::Config.
by aleloi
· 8 years ago
dd31071
Added an empty AudioTransportProxy to AudioState.
by aleloi
· 8 years ago
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
fffa42b
Replace scoped_ptr with unique_ptr in webrtc/audio/
by kwiberg
· 9 years ago
3a94154
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
566ef24
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
by solenberg
· 9 years ago