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gerrit-public.fairphone.software / platform / external / webrtc / 56d460902e1609f7e34d530a463435a71adb2577 / audio
  1. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 8 years ago
  2. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 8 years ago
  3. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 8 years ago
  4. 63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 8 years ago
  5. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 8 years ago
  6. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 8 years ago
  7. c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 8 years ago
  8. 61a7b14 Removing conditional visibility. by Mirko Bonadei · 8 years ago
  9. 6d85252 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up) by henrika · 8 years ago
  10. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 8 years ago
  11. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 8 years ago
  12. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 8 years ago
  13. e4be4b7 Revert "Remove const from ThreadChecker in NullAudioPoller." by Mirko Bonadei · 8 years ago
  14. 54e41dd Remove const from ThreadChecker in NullAudioPoller. by Bjorn Terelius · 8 years ago
  15. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 8 years ago
  16. 9155e49 New classes RefCounter and RefCountedBase. by Niels Möller · 8 years ago
  17. 78609d5 Reland of BWE allocation strategy by Alex Narest · 8 years ago
  18. 6f72f56 Change return types of refcount methods. by Niels Möller · 8 years ago
  19. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 8 years ago
  20. a5fbc23 BWE allocation strategy by Alex Narest · 8 years ago
  21. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 8 years ago
  22. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 8 years ago
  23. b3944f0 Media track ID visibility at BWE level by Alex Narest · 8 years ago
  24. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 8 years ago
  25. 88b23f6 Fix flag name in low_bandwidth_audio_test.py by Edward Lemur · 8 years ago
  26. 7e3b569 Ignore swarming arguments in low_bandwidth_audio_test.py by Edward Lemur · 8 years ago
  27. b0250f0 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 8 years ago
  28. 90e1f53 Fix potentional race in AudioSendStream constructor by Danil Chapovalov · 8 years ago
  29. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 8 years ago
  30. 45a0b36 Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 8 years ago
  31. f4898a6 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 8 years ago
  32. bb1222f Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 8 years ago
  33. 2019698 Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script. by Edward Lemur · 8 years ago
  34. 2011075 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. by Edward Lemur · 8 years ago
  35. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 8 years ago
  36. 1c239d4 Remove voe::Statistics. by solenberg · 8 years ago
  37. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 8 years ago
  38. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 8 years ago
  39. 4652e86 Disable flaky AudioStats.NoLoss test. by solenberg · 8 years ago
  40. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  41. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 8 years ago
  42. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  43. 5a6aa4f Fix path to root in low_bandwidth_audio_test.py by Henrik Kjellander · 8 years ago
  44. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  45. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago