e84978f3d8
Add a Config parameter to AudioProcessing::Create().
...
Also add a parameter-less version; the (int) version is deprecated and
should be removed.
TBR=aluebs,bjornv
BUG=2844
Review URL: https://webrtc-codereview.appspot.com/7609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-25 02:09:06 +00:00
60730cfe3c
Remove the requirement to call set_sample_rate_hz and friends.
...
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org , bjornv@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 17:45:09 +00:00
863b536100
Allow opening an AEC dump from an existing file handle.
...
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.
This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.
BUG=2567
R=andrew@webrtc.org , henrika@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 16:05:17 +00:00
0b72f5863b
Add experimental noise suppression dummy API.
...
Add this flag to the voe_cmd_test.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-19 15:17:51 +00:00
9162080527
Fix some chromium-style warnings in webrtc/modules/audio_processing/
...
BUG=163
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1902004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4472 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-02 11:44:11 +00:00
61e596fc49
Add a Config class interface to AudioProcessing for passing options.
...
Pass the Config down to all AudioProcessing components.
Also add an EchoCancellationImplWrapper to optionally create different
EchoCancellationImpls.
BUG=2117
TBR=turaj@webrtc.org
TESTED=git try
Review URL: https://webrtc-codereview.appspot.com/1843004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4400 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 18:28:29 +00:00
7fad4b8c9f
Include files from webrtc/.. paths in audio_processing/
...
BUG=1662
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4116 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-28 08:11:59 +00:00
b7192b8247
WebRtc_Word32 -> int32_t in audio_processing/
...
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3809 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-10 07:50:54 +00:00
14b43beb7c
Move src/ -> webrtc/
...
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
