Commit Graph

14 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
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
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