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gerrit-public.fairphone.software / platform / external / webrtc / fd5166c305068772d00ad7edf50151bba215400b / call / audio_send_stream.cc
  1. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  2. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  3. c69a56e Remove more unneeded things from ChannelSend by Fredrik Solenberg · 6 years ago
  4. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  5. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  6. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  7. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  8. 0a713b6 replace stringstream in call/ by Jonas Olsson · 7 years ago
  9. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
  10. dee9191 Use rtc::ToString instead of std::to_string by Jiawei Ou · 7 years ago
  11. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  12. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  13. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/audio_send_stream.cc]
  14. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  15. 922246a Replace NULL with nullptr or null in webrtc/audio/ and common_audio/. by deadbeef · 8 years ago
  16. 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]
  17. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  18. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  19. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  20. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  21. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  22. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago