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gerrit-public.fairphone.software / platform / external / webrtc / fc29b0ad4658ccf68a51f2f170ed5641e354bf0a / audio / remix_resample_unittest.cc
  1. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 5 years ago
  2. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  3. f0b8dee Qualify cmath functions. by Mirko Bonadei · 6 years ago
  4. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  5. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  6. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago[Renamed (98%) from voice_engine/utility_unittest.cc]
  7. 55900fd Move APM initialization into WebRtcVoiceEngine by Fredrik Solenberg · 7 years ago
  8. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  9. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/voice_engine/utility_unittest.cc]
  10. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  11. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  12. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  13. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 7 years ago
  14. 228c268 Support 4 channel mic in Windows Core Audio by jens.nielsen · 8 years ago
  15. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  16. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  17. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  18. cdfe20b Fix the maximum native sample rate in AudioProcessing by Alejandro Luebs · 9 years ago
  19. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  20. b7e5054 Match existing type usage better. by Peter Kasting · 9 years ago
  21. f5a33f1 Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
  22. a78a41f Move output_mixer_unittest.cc to utility_unittest.cc. by andrew@webrtc.org · 11 years ago[Renamed (96%) from webrtc/voice_engine/output_mixer_unittest.cc]
  23. 40ee3d0 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 11 years ago
  24. c1eb560 Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
  25. 50b2efe Add a wrapper around PushSincResampler and the old Resampler. by andrew@webrtc.org · 11 years ago
  26. ae1a58b Replace AudioFrame's operator= with CopyFrom(). by andrew@webrtc.org · 12 years ago
  27. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/voice_engine/output_mixer_unittest.cc]
  28. 6f8db36 Reorganize voice_engine/. by andrew@webrtc.org · 12 years ago[Renamed from src/voice_engine/main/source/output_mixer_unittest.cc]
  29. 4ecea3e Downmix before resampling in capture and render paths. by andrew@webrtc.org · 12 years ago