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16 Commits

Author SHA1 Message Date
pbos@webrtc.org
e468bc9e60 Rename _t struct types in audio_processing. 
_t names are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:11:33 +00:00
aluebs@webrtc.org
cf6d0b64ef Add 48kHz support to AGC 
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

Originally reviewed here:
https://webrtc-codereview.appspot.com/26339004/

BUG=webrtc:3146
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7917 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 20:56:09 +00:00
bjornv@webrtc.org
b395a5ea65 audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/ 
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.

BUG=
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 10:38:10 +00:00
aluebs@webrtc.org
c5ebbd98f5 Support 48kHz in Noise Suppression 
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 19:30:57 +00:00
aluebs@webrtc.org
a7384a1126 Simplify audio_buffer APIs 
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 01:06:35 +00:00
aluebs@webrtc.org
2561d52460 Simplify AudioBuffer::mixed_low_pass_data API 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:27:39 +00:00
andrew@webrtc.org
65f933899b Fix constness of AudioBuffer accessors. 
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.

Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 16:44:13 +00:00
andrew@webrtc.org
ddbb8a2c24 Support arbitrary input/output rates and downmixing in AudioProcessing. 
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.

- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 21:00:04 +00:00
bjornv@webrtc.org
5964fe0f86 audio_processing: DestroyHandle() now returns void 
The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.

BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-22 06:52:28 +00:00
andrew@webrtc.org
56e4a05053 Remove ProcessingComponent's dependence on AudioProcessingImpl. 
- Move needed accessors to AudioProcessing.
- Inject the crit directly as a dependency.
- Remove the now unneeded EchoCancellationImplWrapper.

BUG=2894
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 22:23:17 +00:00
andrew@webrtc.org
b6541ca3a1 Ensure capture_levels_ is sized correctly at init time. 
Fixes failing voe_auto_test and audioproc_perf.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:36:10 +00:00
andrew@webrtc.org
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
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
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
andrew@webrtc.org
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