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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
de6fa1ef2946002c305861742561f94b8606426d
/
call
/
audio_state.h
d2c336f
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
by Henrik Boström
· 5 years ago
d970807
Remove rtc_base/scoped_ref_ptr.h.
by Mirko Bonadei
· 6 years ago
921d366
Remove comments about using std::shared_ptr.
by Mirko Bonadei
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
11b34f4
Remove chromium clang style errors affecting sdk/android/media_jni
by Paulina Hensman
· 6 years ago
f120cba
Delete AudioMonitor and related code.
by Niels Möller
· 7 years ago
a8b7c7f
Move remaining traces of VoiceEngine
by Fredrik Solenberg
· 7 years ago
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
cf73c96
Add AudioDeviceModule to AudioState::Config.
by Fredrik Solenberg
· 7 years ago
63e6072
Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.
by Fredrik Solenberg
· 7 years ago
5f6bf24
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)
by henrika
· 7 years ago
990d6b8
Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
by Mirko Bonadei
· 7 years ago
90bace0
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
by henrika
· 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/call/audio_state.h]
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 7 years ago
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]
16e3caa
Removed unused forward declaration.
by aleloi
· 8 years ago
81da488
Added audio mixer and removed audio device module in AudioState::Config.
by aleloi
· 8 years ago
a69d973
Move webrtc/audio_*.h to webrtc/api/call
by kjellander
· 8 years ago
[Renamed (92%) from webrtc/audio_state.h]
a4527c8
Add comments about the Audio parts of the public Call API being WIP.
by Fredrik 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