1. 382c0c2 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 11 years ago
  2. e44a84d Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 11 years ago
  3. 82a045a APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 11 years ago
  4. 46b31b1 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 11 years ago
  5. ddbb8a2 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 11 years ago
  6. 40ee3d0 Consolidate audio conversion from Channel and TransmitMixer. by andrew@webrtc.org · 11 years ago
  7. a8b9737 Add tests and modify tools for new float deinterleaved interface. by andrew@webrtc.org · 11 years ago
  8. 17e4064 Add a deinterleaved float interface to AudioProcessing. by andrew@webrtc.org · 11 years ago
  9. 56e4a05 Remove ProcessingComponent's dependence on AudioProcessingImpl. by andrew@webrtc.org · 11 years ago
  10. f92aaff AudioProcessing is not a Module. by andrew@webrtc.org · 11 years ago
  11. 17342e5 Add a method to inform AudioProcessing that its output will be muted. by andrew@webrtc.org · 11 years ago
  12. 75dd288 Add an interface for accepting keypress signals to AudioProcessing. by andrew@webrtc.org · 11 years ago
  13. 5474491 Update AudioProcessing::Create docs. by andrew@webrtc.org · 11 years ago
  14. c7c7a53 Add Config struct for experimental AGC. by andrew@webrtc.org · 11 years ago
  15. e84978f Add a Config parameter to AudioProcessing::Create(). by andrew@webrtc.org · 11 years ago
  16. 60730cf Remove the requirement to call set_sample_rate_hz and friends. by andrew@webrtc.org · 12 years ago
  17. 863b536 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 12 years ago
  18. 0b72f58 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 12 years ago
  19. 6b1e219 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 12 years ago
  20. f3930e9 Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 12 years ago
  21. 9162080 Fix some chromium-style warnings in webrtc/modules/audio_processing/ by pbos@webrtc.org · 12 years ago
  22. 61e596f Add a Config class interface to AudioProcessing for passing options. by andrew@webrtc.org · 12 years ago
  23. b7192b8 WebRtc_Word32 -> int32_t in audio_processing/ by pbos@webrtc.org · 12 years ago
  24. 91d11b3 Adds new AEC API to audio_processing. by bjornv@webrtc.org · 12 years ago
  25. 6be1e93 Properly error check calls to AudioProcessing. by andrew@webrtc.org · 12 years ago
  26. d72b3d6 Fix cpplint errors in audio_processing.h by andrew@webrtc.org · 13 years ago
  27. 8186534 Only reinitialize AudioProcessing when needed. by andrew@webrtc.org · 13 years ago
  28. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 13 years ago[Renamed from src/modules/audio_processing/include/audio_processing.h]
  29. 08329f4 Added API to port internal speech probability in NS. by bjornv@webrtc.org · 13 years ago
  30. 63a5098 Rename AudioFrame members. by andrew@webrtc.org · 13 years ago
  31. 6f9f817 Add an API to offset system delay. by andrew@webrtc.org · 13 years ago
  32. 648af74 Clean up MapSetting(). by andrew@webrtc.org · 13 years ago
  33. b9d7d93 Rename interface/ to include/ in audio_processing. by andrew@webrtc.org · 13 years ago[Renamed from src/modules/audio_processing/interface/audio_processing.h]
  34. 5ae19de Fix error in RtpDump::Start due to r1156. by andrew@webrtc.org · 14 years ago
  35. 755b04a Add RMS computation for the RTP level indicator. by andrew@webrtc.org · 14 years ago
  36. c4f129f Improve the mixing saturation protection scheme. by andrew@webrtc.org · 14 years ago
  37. 4d5d5c1 Reorganize the audio_processing source. by andrew@webrtc.org · 14 years ago[Renamed from src/modules/audio_processing/main/interface/audio_processing.h]
  38. 1ba3dbe Adds possibility to log delay estimates in AEC. by bjornv@google.com · 14 years ago
  39. ed083d4 Modify the _vadActivity member of the AudioFrame passed to AudioProcessing. by andrew@webrtc.org · 14 years ago
  40. 22e6515 Changing echo_path_size_bytes() to static, and using size_t rather than int. This is recommended by Chromium: by ajm@google.com · 14 years ago
  41. c4b939c Added calls to set and get external echo channels. by bjornv@google.com · 14 years ago
  42. 470e71d by niklase@google.com · 14 years ago