Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / fc29b0ad4658ccf68a51f2f170ed5641e354bf0a / audio / audio_transport_impl.h
  1. b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 4 years, 8 months ago
  2. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  3. d970807 Remove rtc_base/scoped_ref_ptr.h. 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 · 6 years ago
  5. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  6. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  7. 649a385 Removes usage of analog AGC. by henrika · 7 years ago
  8. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago