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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
/
include
/
audio_processing.h
382c0c2
Allow the RTP level indicator computation to work at any sample rate.
by andrew@webrtc.org
· 11 years ago
e44a84d
Only clamp to 16 kHz when AECM is enabled.
by andrew@webrtc.org
· 11 years ago
82a045a
APM: limit native sample rate to 16kHz on mobile.
by fischman@webrtc.org
· 11 years ago
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
40ee3d0
Consolidate audio conversion from Channel and TransmitMixer.
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
5474491
Update AudioProcessing::Create docs.
by andrew@webrtc.org
· 11 years ago
c7c7a53
Add Config struct for experimental AGC.
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
6b1e219
Move the Config DelayCorrection struct to audio_processing.h.
by andrew@webrtc.org
· 12 years ago
f3930e9
Small refactoring of AudioProcessing use in channel.cc.
by andrew@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
b7192b8
WebRtc_Word32 -> int32_t in audio_processing/
by pbos@webrtc.org
· 12 years ago
91d11b3
Adds new AEC API to audio_processing.
by bjornv@webrtc.org
· 12 years ago
6be1e93
Properly error check calls to AudioProcessing.
by andrew@webrtc.org
· 12 years ago
d72b3d6
Fix cpplint errors in audio_processing.h
by andrew@webrtc.org
· 13 years ago
8186534
Only reinitialize AudioProcessing when needed.
by andrew@webrtc.org
· 13 years ago
14b43be
Move src/ -> webrtc/
by andrew@webrtc.org
· 13 years ago
[Renamed from src/modules/audio_processing/include/audio_processing.h]
08329f4
Added API to port internal speech probability in NS.
by bjornv@webrtc.org
· 13 years ago
63a5098
Rename AudioFrame members.
by andrew@webrtc.org
· 13 years ago
6f9f817
Add an API to offset system delay.
by andrew@webrtc.org
· 13 years ago
648af74
Clean up MapSetting().
by andrew@webrtc.org
· 13 years ago
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]
5ae19de
Fix error in RtpDump::Start due to r1156.
by andrew@webrtc.org
· 14 years ago
755b04a
Add RMS computation for the RTP level indicator.
by andrew@webrtc.org
· 14 years ago
c4f129f
Improve the mixing saturation protection scheme.
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/interface/audio_processing.h]
1ba3dbe
Adds possibility to log delay estimates in AEC.
by bjornv@google.com
· 14 years ago
ed083d4
Modify the _vadActivity member of the AudioFrame passed to AudioProcessing.
by andrew@webrtc.org
· 14 years ago
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
c4b939c
Added calls to set and get external echo channels.
by bjornv@google.com
· 14 years ago
470e71d
by niklase@google.com
· 14 years ago