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gerrit-public.fairphone.software / platform / external / webrtc / 56da2f78687d311111679c089cc639788a5b3dec / audio
  1. 3c24ea8 Removed SetTransportOverhead in transport controller. by Sebastian Jansson · 7 years ago
  2. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
  3. f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
  4. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  5. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  6. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  7. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  8. 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
  9. 4c1ffb8 Removing access to pacer in rtp controller. by Sebastian Jansson · 7 years ago
  10. e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 7 years ago
  11. 1e06289 Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. by Niels Möller · 7 years ago
  12. dbbb33c Stop using public_deps in common_audio. by Mirko Bonadei · 7 years ago
  13. 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
  14. ed7b4ff Use isolated-script-test-perf-output on low_bandwidth_audio_test. by Edward Lemur · 7 years ago
  15. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  16. 75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
  17. 001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
  18. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  19. 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
  20. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  21. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  22. d8b041c Ignore extra arguments in low_bandwidth_audio_test. by Edward Lemur · 7 years ago
  23. 649a385 Removes usage of analog AGC. by henrika · 7 years ago
  24. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  25. 98d4036 Make it possible to run low_bandwidth_audio_test on Android swarming. by Edward Lemur · 7 years ago
  26. b401771 Store JSON perf results for low_bandwidth_audio_test. by Edward Lemur · 7 years ago
  27. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  28. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  29. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  30. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  31. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  32. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  33. 731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  34. 5a25ab2 Revert "Add mock_rtc_event_log.h." by Edward Lemur · 7 years ago
  35. 63aea46 Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  36. 94dc177 Add mock_bitrate_controller.h. by Patrik Höglund · 7 years ago
  37. 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
  38. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  39. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  40. f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago
  41. d524751 Replace VoEBase::[Start/Stop]Playout(). by Fredrik Solenberg · 7 years ago
  42. aaedf75 Replace VoEBase::[Start/Stop]Send(). by Fredrik Solenberg · 7 years ago
  43. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  44. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  45. a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  46. d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
  47. a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  48. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  49. b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
  50. 5e849cf Stop using public_deps in audio/utility. by Mirko Bonadei · 7 years ago
  51. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  52. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  53. d319534 Move ADM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  54. 63e6072 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. by Fredrik Solenberg · 7 years ago
  55. 2707fb2 Optional: Use nullopt and implicit construction in /audio by Oskar Sundbom · 7 years ago
  56. 8d9c540 Deprecated BitrateController::CreateRtcpBandwidthObserver. by Sebastian Jansson · 7 years ago
  57. c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
  58. 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
  59. 6d85252 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection AP (follow-up) by henrika · 7 years ago
  60. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  61. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  62. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  63. e4be4b7 Revert "Remove const from ThreadChecker in NullAudioPoller." by Mirko Bonadei · 7 years ago
  64. 54e41dd Remove const from ThreadChecker in NullAudioPoller. by Bjorn Terelius · 7 years ago
  65. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  66. 9155e49 New classes RefCounter and RefCountedBase. by Niels Möller · 7 years ago
  67. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  68. 6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
  69. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  70. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  71. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  72. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  73. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  74. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  75. 88b23f6 Fix flag name in low_bandwidth_audio_test.py by Edward Lemur · 7 years ago
  76. 7e3b569 Ignore swarming arguments in low_bandwidth_audio_test.py by Edward Lemur · 7 years ago
  77. b0250f0 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
  78. 90e1f53 Fix potentional race in AudioSendStream constructor by Danil Chapovalov · 7 years ago
  79. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  80. 45a0b36 Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
  81. f4898a6 Reland "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
  82. bb1222f Revert "Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script." by Edward Lemur · 7 years ago
  83. 2019698 Don't download PESQ and POLQA in the low_bandwidth_audio_test.py script. by Edward Lemur · 7 years ago
  84. 2011075 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. by Edward Lemur · 7 years ago
  85. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  86. 1c239d4 Remove voe::Statistics. by solenberg · 7 years ago
  87. fc3a2e3 Remove the VoiceEngineObserver callback interface. by solenberg · 7 years ago
  88. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  89. 4652e86 Disable flaky AudioStats.NoLoss test. by solenberg · 7 years ago
  90. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  91. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
  92. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  93. 5a6aa4f Fix path to root in low_bandwidth_audio_test.py by Henrik Kjellander · 7 years ago
  94. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  95. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago