1. 3aa28f0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  2. 5b18967 Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  3. 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  4. ea04cc0 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  5. d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 7 years ago
  6. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  7. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  8. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  9. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  10. fe5f71c Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 8 years ago
  11. ee5a316 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  12. 31212ba Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (96%) from api/call/audio_send_stream.h]
  13. 017ebe5 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  14. 9a1d49f Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  15. ff9d77c Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  16. a1912ef Fixing config for Audio BWE. by minyue · 8 years ago
  17. 5206602 Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  18. 91c6f34 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  19. 77b3ea6 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  20. b8bff80 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  21. dfb640f Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  22. bffc190 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  23. cf82062 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (95%) from audio_send_stream.h]
  24. f29ccf6 Variable audio bitrate. by mflodman · 8 years ago
  25. d9cd888 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 9 years ago
  26. 5a37e3e Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 9 years ago
  27. 8f99654 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 9 years ago
  28. bcf3191 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  29. 97aa5c2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 9 years ago
  30. 8b348aa Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  31. 47a40a3 Remove webrtc/stream.h and unutilized inheritance. by pbos · 9 years ago
  32. b670f85 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 9 years ago
  33. f2e3315 Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 9 years ago
  34. b9a65af Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 9 years ago
  35. 9ef75db - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 9 years ago
  36. f95302f Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  37. ae4b1f0 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  38. f1f7cbb Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  39. ffe1ce0 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  40. 1c28f5c Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  41. f707c68 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  42. 0c670ff Align new VoE API with design. by solenberg · 9 years ago
  43. d5bdda3 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
  44. dce9bf3 Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  45. 2fb88e4 Define Stream base classes by Jelena Marusic · 9 years ago
  46. ee9b72e VoE2 API draft by Fredrik Solenberg · 10 years ago