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19 Commits

Author SHA1 Message Date
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
andrew@webrtc.org
22858d4785 Add an extended filter option to audioproc. 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 14:07:17 +00:00
andrew@webrtc.org
ca764ab22d Add a parameter to audioproc for overriding the delay. 
Rename the parameter for adding to the input delay to "add_delay".

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2345007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-07 16:44:32 +00:00
andrew@webrtc.org
f3930e941c Small refactoring of AudioProcessing use in channel.cc. 
- Apply consistent naming.
- Use a scoped_ptr for rx_audioproc_.
- Remove now unnecessary AudioProcessing::Destroy().

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

Review URL: https://webrtc-codereview.appspot.com/2184007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4784 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 22:37:32 +00:00
pbos@webrtc.org
8c34ceeef1 Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD 
BUG=
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1571004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 09:24:03 +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
andrew@webrtc.org
dff69c56b0 Add AEC suppression level option to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1368007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 23:01:09 +00:00
andrew@webrtc.org
1acb3b33bc Add comfort noise disabling and routing mode selection to audioproc. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1358004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-26 00:39:27 +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
6be1e934ad Properly error check calls to AudioProcessing. 
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.

BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 18:47:28 +00:00
kjellander@webrtc.org
00ab7cf4fd Fix perf output for audioproc and iSAC fixed-point tests 
The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.

BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/1093007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-11 12:33:03 +00:00
andrew@webrtc.org
bafdae3cfc Fix simulated analog gain in audioproc. 
* It doesn't make much sense to apply at all when reading from the protobuf.
* Reduced the gain to be closer to actual mics.

BUG=1260

Review URL: https://webrtc-codereview.appspot.com/1027007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 23:11:29 +00:00
kjellander@webrtc.org
10abe25f6d Make audioproc output files be written to output dir by default. 
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat

TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1003005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-17 18:28:07 +00:00
kma@webrtc.org
0e739508e0 Added buildbot benchmarking in iSAC and APM into Android platform build. 
Review URL: https://webrtc-codereview.appspot.com/964022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-07 15:26:28 +00:00
andrew@webrtc.org
b43502e388 Revert 3170 - Added performance benchmarking in APM and iSAC-fix for Buildbots. 
Review URL: https://webrtc-codereview.appspot.com/929022

TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 23:57:38 +00:00
kma@webrtc.org
4cd8f1f182 Added performance benchmarking in APM and iSAC-fix for Buildbots. 
Review URL: https://webrtc-codereview.appspot.com/929022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-26 22:02:47 +00:00
andrew@webrtc.org
8186534111 Only reinitialize AudioProcessing when needed. 
This takes away the burden from the user, resulting in cleaner code.

Review URL: https://webrtc-codereview.appspot.com/941005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3010 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-27 00:28:27 +00:00
leozwang@webrtc.org
534e495df0 Qickly fixed android platform build breakage 
TBR=ajm
Review URL: https://webrtc-codereview.appspot.com/920006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-22 21:21:52 +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