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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
fe3dfeeb3c93e7d9a9df9461150c32c03b4925ac
/
audio
/
audio_state.cc
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
ac63ac7
Update refcounting of AudioState to use rtc::RefCountedObject
by Niels Möller
· 6 years ago
8fdcac3
Remove clang:find_bad_constructs suppression from call:call.
by Mirko Bonadei
· 6 years ago
918f50c
Use absl::make_unique and absl::WrapUnique directly
by Karl Wiberg
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
f120cba
Delete AudioMonitor and related code.
by Niels Möller
· 7 years ago
24ea822
Remove logging in audio/* from release builds.
by Jonas Olsson
· 7 years ago
649a385
Removes usage of analog AGC.
by henrika
· 7 years ago
8f5787a
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
by Fredrik Solenberg
· 7 years ago
d524751
Replace VoEBase::[Start/Stop]Playout().
by Fredrik Solenberg
· 7 years ago
aaedf75
Replace VoEBase::[Start/Stop]Send().
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
6d85252
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up)
by henrika
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 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
6f72f56
Change return types of refcount methods.
by Niels Möller
· 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.cc]
e67bedb
External APM usage downstream dependency support cleanup
by peah
· 7 years ago
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
10111bc
Passed AudioMixer to AudioState::Config.
by aleloi
· 8 years ago
dd31071
Added an empty AudioTransportProxy to AudioState.
by aleloi
· 8 years ago
566ef24
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
by solenberg
· 9 years ago