1. c848268 Use SequenceChecker(SequenceChecker::kDetached) in a few places. by Tommi · 1 year, 10 months ago
  2. 0cf140d Rewrite AudioState null poller to use TaskQueueBase interface by Danil Chapovalov · 2 years, 5 months ago
  3. 105711e Move rtc::make_ref_counted to api/ by Niels Möller · 2 years, 7 months ago
  4. 3176ef7 Rename AudioReceiveStream to AudioReceiveStreamInterface by Tommi · 2 years, 8 months ago
  5. dddbbeb Rename internal::AudioReceiveStream to AudioReceiveStreamImpl by Tommi · 2 years, 8 months ago
  6. ba2de58 Update audio/, media/, and video/ to not use implicit conversion by Niels Möller · 2 years, 9 months ago
  7. d7fdb95 Remove typing detection by Alessio Bazzica · 2 years, 10 months ago
  8. 97597c0 Remove usage of INFO alias for LS_INFO in log messages by Harald Alvestrand · 3 years, 3 months ago
  9. c1d5891 Replace `new rtc::RefCountedObject` with `rtc::make_ref_counted` in a few files by Tomas Gunnarsson · 3 years, 9 months ago
  10. 09ceed2 Async audio processing API by Olga Sharonova · 4 years, 4 months ago
  11. cc73ed3 APM: Add build flag to allow building WebRTC without APM by Per Åhgren · 4 years, 9 months ago
  12. fdbbada Revert "Inlines NullAudioPoller functionality into AudioState class." by Sebastian Jansson · 5 years ago
  13. 0e96535 Inlines NullAudioPoller functionality into AudioState class. by Sebastian Jansson · 5 years ago
  14. b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 5 years ago
  15. dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 5 years ago
  16. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
  17. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 6 years ago
  18. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
  19. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  20. ac63ac7 Update refcounting of AudioState to use rtc::RefCountedObject by Niels Möller · 6 years ago
  21. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  22. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 7 years ago
  23. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  24. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  25. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  26. 649a385 Removes usage of analog AGC. by henrika · 7 years ago
  27. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  28. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  29. aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
  30. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  31. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  32. 6d85252 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up) by henrika · 7 years ago
  33. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  34. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  35. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  36. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  37. 6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
  38. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  39. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  40. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/audio_state.cc]
  41. e67bedb External APM usage downstream dependency support cleanup by peah · 8 years ago
  42. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  43. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  44. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  45. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  46. 10111bc Passed AudioMixer to AudioState::Config. by aleloi · 8 years ago
  47. dd31071 Added an empty AudioTransportProxy to AudioState. by aleloi · 8 years ago
  48. 566ef24 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). by solenberg · 9 years ago