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Author SHA1 Message Date
andrew@webrtc.org
27c6980239 Move the volume quantization workaround from VoE to AGC. 
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 20:24:56 +00:00
andrew@webrtc.org
ce8e077cf0 Add a keypress field to the audioproc debug proto. 
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.

TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.

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

Review URL: https://webrtc-codereview.appspot.com/8509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 15:28:30 +00:00
aluebs@webrtc.org
c9ee412070 Re-enabling audio processing tests 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 14:41:57 +00:00
bjornv@webrtc.org
6a94734d4d Adds back set_sample_rate_hz() when Init is called in recordings. 
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.

BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 08:41:09 +00:00
aluebs@webrtc.org
8bc4fcfeb6 Temporarily disabling audio processing tests. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6889005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:14:47 +00:00
bjornv@webrtc.org
bbd47fc5b5 Enables robust delay validation in AEC delay logging. 
* Explicitly disabled robust validation in AECM.
* Updated audio_processing_unittests for using robust delay validation in AEC.
* Updated output_data_float.pb (not needed for Android nor fixed point, since AECM is untouched).

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 08:54:34 +00:00
andrew@webrtc.org
d335094852 Init to 16 kHz in the fixed-point profile. 
Fixes modules_unittests for fixed-point builds (Android).

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:57:10 +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
fischman@webrtc.org
f8be8df33a audio_processing_unittest: unbreak clang compilation. 
BUG=2735
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 23:46:39 +00:00
henrikg@webrtc.org
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
andrew@webrtc.org
3d9981d58a Remove unused ThreadData struct. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/4949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 17:13:47 +00:00
andrew@webrtc.org
d7696c4ed1 Compile-out functions only used by the bit-exact test. 
Causes errors on platforms where the test is unused.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/4869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 23:39:16 +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
kjellander@webrtc.org
3555303cb0 Roll chromium_revision 226126:228675 and fix clang warnings 
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.

This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.

TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 20:10:17 +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
henrike@webrtc.org
a950300b0e Disables unit tests that don't work on Android for Android. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 18:53:54 +00:00
pbos@webrtc.org
c66aaaf921 Rename unit_test.{cc,h} under module_unittest. 
Squelches the following Windows trybot warning:
warning LNK4042: object specified more than once; extras ignored

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

Review URL: https://webrtc-codereview.appspot.com/1758004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 07:56:33 +00:00
henrike@webrtc.org
83cebb25d7 Removes unused main function that is poluting the build. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1728005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 18:31:13 +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
bjornv@webrtc.org
91d11b3cdd Adds new AEC API to audio_processing. 
One unit test added.
Tested with audioproc_unittest and trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 16:53:09 +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
andrew@webrtc.org
78693fe37c Return an error when greater than 16 kHz is used with AECM. 
BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1146005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 16:36:19 +00:00
bjornv@webrtc.org
3e10249f20 Added delay estimation test to audio processing unit tests. 
The test verifies that we get proper delay metrics when inserting delayed versions of the same file to far-end and near-end.
Failure of the test has been verified through a missmatch between AEC delay buffer size and test buffer size.
Also added a missing file rewind to another test and removed some lint warnings.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1100004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-14 15:29:09 +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
ae1a58bba4 Replace AudioFrame's operator= with CopyFrom(). 
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.

Review URL: https://webrtc-codereview.appspot.com/1031007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 04:44:30 +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