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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
fc29b0ad4658ccf68a51f2f170ed5641e354bf0a
/
audio
/
audio_transport_impl.h
b8c775a
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
by Tim Na
· 4 years, 8 months ago
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
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
a8b7c7f
Move remaining traces of VoiceEngine
by Fredrik Solenberg
· 7 years ago
649a385
Removes usage of analog AGC.
by henrika
· 7 years ago
2a87797
Remove voe::TransmitMixer
by Fredrik Solenberg
· 7 years ago