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webrtc.googlesource.com
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src
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9c55f0f957534144d2b8a64154f0a479249b34be
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webrtc
/
modules
/
audio_processing
/
audio_processing_impl.h
46b31b1
Restore sample_rate_hz() until Chromium is updated to not use it.
by andrew@webrtc.org
· 11 years ago
ddbb8a2
Support arbitrary input/output rates and downmixing in AudioProcessing.
by andrew@webrtc.org
· 11 years ago
a8b9737
Add tests and modify tools for new float deinterleaved interface.
by andrew@webrtc.org
· 11 years ago
17e4064
Add a deinterleaved float interface to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
56e4a05
Remove ProcessingComponent's dependence on AudioProcessingImpl.
by andrew@webrtc.org
· 11 years ago
f92aaff
AudioProcessing is not a Module.
by andrew@webrtc.org
· 11 years ago
17342e5
Add a method to inform AudioProcessing that its output will be muted.
by andrew@webrtc.org
· 11 years ago
75dd288
Add an interface for accepting keypress signals to AudioProcessing.
by andrew@webrtc.org
· 11 years ago
e84978f
Add a Config parameter to AudioProcessing::Create().
by andrew@webrtc.org
· 11 years ago
60730cf
Remove the requirement to call set_sample_rate_hz and friends.
by andrew@webrtc.org
· 12 years ago
863b536
Allow opening an AEC dump from an existing file handle.
by henrikg@webrtc.org
· 12 years ago
0b72f58
Add experimental noise suppression dummy API.
by aluebs@webrtc.org
· 12 years ago
9162080
Fix some chromium-style warnings in webrtc/modules/audio_processing/
by pbos@webrtc.org
· 12 years ago
61e596f
Add a Config class interface to AudioProcessing for passing options.
by andrew@webrtc.org
· 12 years ago
7fad4b8
Include files from webrtc/.. paths in audio_processing/
by pbos@webrtc.org
· 12 years ago
b7192b8
WebRtc_Word32 -> int32_t in audio_processing/
by pbos@webrtc.org
· 12 years ago
14b43be
Move src/ -> webrtc/
by andrew@webrtc.org
· 13 years ago
[Renamed from src/modules/audio_processing/audio_processing_impl.h]
369166a
Add API for disabling the high pass filter.
by andrew@webrtc.org
· 13 years ago
6f9f817
Add an API to offset system delay.
by andrew@webrtc.org
· 13 years ago
c450a19
Removed Version function from all modules.
by pwestin@webrtc.org
· 14 years ago
7bf2646
Make protobuf use optional.
by andrew@webrtc.org
· 14 years ago
755b04a
Add RMS computation for the RTP level indicator.
by andrew@webrtc.org
· 14 years ago
4d5d5c1
Reorganize the audio_processing source.
by andrew@webrtc.org
· 14 years ago
[Renamed from src/modules/audio_processing/main/source/audio_processing_impl.h]
808e0e0
Update the debug recordings to use protobufs.
by ajm@google.com
· 14 years ago
470e71d
by niklase@google.com
· 14 years ago