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gerrit-public.fairphone.software / platform / external / webrtc / 92ea95e34af5966555903026f45164afbd7e2088 / api / call / audio_sink.h
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/call/audio_sink.h]
  3. 0acebe2 Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ ) by zhihuang · 7 years ago
  4. c904634 Revert of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2873803002/ ) by zhihuang · 7 years ago
  5. 3860596 Make AudioSinkInterface::Data hold a const pointer to the audio buffer. by yujo · 7 years ago
  6. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (92%) from webrtc/audio_sink.h]
  7. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago[Renamed (92%) from webrtc/audio/audio_sink.h]
  8. 7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago[Renamed (92%) from webrtc/audio_sink.h]
  9. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago[Renamed (92%) from webrtc/audio/audio_sink.h]
  10. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  11. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  12. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  13. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago