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gerrit-public.fairphone.software / platform / external / webrtc / 33fe1dfd3a03d8a6fee8e12bbc0093bc0360d0eb / call / audio_send_stream.cc
  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/call/audio_send_stream.cc]
  3. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  4. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  5. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (98%) from webrtc/api/call/audio_send_stream.cc]
  6. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  7. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  8. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  9. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  10. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  11. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago