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gerrit-public.fairphone.software / platform / external / webrtc / c3d0da097cd5040c6108d099f0291b3e138db0cb / call / audio_send_stream.h
  1. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  2. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  3. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/audio_send_stream.h]
  4. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  5. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  6. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  7. abbc430 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public. by eladalon · 7 years ago
  8. e76bd3a 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
  9. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  10. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  11. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  12. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  13. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  14. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  15. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago[Renamed (96%) from webrtc/api/call/audio_send_stream.h]
  16. a8eb756 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. by aleloi · 8 years ago
  17. 1acfbd2 Expose RtpCodecParameters to VoiceMediaInfo stats. by hbos · 8 years ago
  18. ffbbcac Support multiple timestamp rates for sending DTMF. by solenberg · 8 years ago
  19. 10cbb46 Fixing config for Audio BWE. by minyue · 8 years ago
  20. 6b825df Using AudioOption to enable audio network adaptor. by minyue · 8 years ago
  21. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  22. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  23. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  24. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  25. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  26. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago[Renamed (96%) from webrtc/audio_send_stream.h]
  27. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  28. 9421853 Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream(). by solenberg · 8 years ago
  29. 971cab0 Configure VoE NACK through AudioSendStream::Config, for send streams. by solenberg · 8 years ago
  30. 6806136 Remove RED support from WebRtcVoiceEngine/MediaChannel by kwiberg · 8 years ago
  31. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  32. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 8 years ago
  33. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  34. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  35. bfefb03 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 8 years ago
  36. 8842c3e Relanding https://codereview.webrtc.org/1715883002/ in pieces. by solenberg · 8 years ago
  37. 3ecb5c8 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ ) by solenberg · 8 years ago
  38. 8886c81 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. by solenberg · 8 years ago
  39. b86d4e4 Prepare the AudioSendStream to be hooked up to send-side BWE. by Stefan Holmer · 9 years ago
  40. b572768 - Remove calls to VoEDtmf from WVoE/MC. by Fredrik Solenberg · 9 years ago
  41. a4527c8 Add comments about the Audio parts of the public Call API being WIP. by Fredrik Solenberg · 9 years ago
  42. 3a94154 Move some send stream configuration into webrtc::AudioSendStream. by solenberg · 9 years ago
  43. 85a0496 Implement AudioSendStream::GetStats(). by solenberg · 9 years ago
  44. c7a8b08 Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. by solenberg · 9 years ago
  45. cf18b34 Align new VoE API with design. by solenberg · 9 years ago
  46. 2d56668 Unify Transport and newapi::Transport interfaces. by pbos · 9 years ago
  47. 4fbae2b Add send transports to individual webrtc::Call streams. by solenberg · 9 years ago
  48. cd67022 Define Stream base classes by Jelena Marusic · 9 years ago
  49. 04f4931 VoE2 API draft by Fredrik Solenberg · 9 years ago