behavior of the audio processing module is quite complex and hard to
implement in a threadsafe and efficient manner. Therefore a new
scheme for setting the parameters in the audio processing module is
introduced in this CL.
The idea is to roll this scheme out gradually and as a first functionality
in the audio processing module where this is applied the level controller
was chosen. This CL includes the replacement of the Config-based
level controller scheme with the new scheme.
TBR=henrik.lundin@webrtc.org, solenberg@webrtc.org,
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2338493002
Cr-Commit-Position: refs/heads/master@{#14190}
Reason for revert:
Interface change in the mock breaks downstream code.
Original issue's description:
> The current scheme for setting parameters and specifying the behavior
> of the audio processing module is quite complex and hard to implement
> in a threadsafe and efficient manner. Therefore a new scheme for setting
> the parameters in the audio processing module is introduced in this CL.
>
> The idea is to roll this scheme out gradually and as a first functionality
> in the audio processing module where this is applied the level controller
> was chosen. This CL includes the replacement of the Config-based
> level controller scheme with the new scheme.
>
> BUG=webrtc:5298
>
> Committed: https://crrev.com/c8bbe3fe9aad9e9a1189a42dcaa8f5d6c261ecc8
> Cr-Commit-Position: refs/heads/master@{#14171}
TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org
BUG=webrtc:5298
NOTRY=True
Review-Url: https://codereview.webrtc.org/2334583002
Cr-Commit-Position: refs/heads/master@{#14177}
of the audio processing module is quite complex and hard to implement
in a threadsafe and efficient manner. Therefore a new scheme for setting
the parameters in the audio processing module is introduced in this CL.
The idea is to roll this scheme out gradually and as a first functionality
in the audio processing module where this is applied the level controller
was chosen. This CL includes the replacement of the Config-based
level controller scheme with the new scheme.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2292863002
Cr-Commit-Position: refs/heads/master@{#14171}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
Also added IntelligibilityEnhancer setting to aecdump simulator in audioproc_f
Review-Url: https://codereview.webrtc.org/2075093003
Cr-Commit-Position: refs/heads/master@{#13220}
Some aecdumps have more than one INIT event. In those cases only the last wav file was unpacked, which sometimes is not the most interesting or desired one.
This CL creates a different wav file after each INIT event.
Review-Url: https://codereview.webrtc.org/2067423002
Cr-Commit-Position: refs/heads/master@{#13196}
tool to support all the functionality needed for simulating
and analyzing the audio processing module behavior during
calls.
BUG=
Review-Url: https://codereview.webrtc.org/1907223003
Cr-Commit-Position: refs/heads/master@{#12882}
There were a series of changes in the calculation of echo metrics. There changes made the existing unittests lose, e.g., EXPECT_EQ become EXPECT_NEAR. It is good time to protect the echo calculation more strictly.
The change is not simply generating a new reference file and change EXPECT_NEAR to EXPECT_EQ. It strengthens the test as well. Main changes are
1. the old test only sample a metric at the end of processing, while the new test takes metrics during the call with a certain time interval. This gives a much stronger protection.
2. added protection of a newly added metric, called divergent_filter_fraction.
3. as said, use EXPECT_EQ (actually ASSERT_EQ) instead of EXPECT_NEAR as much as possible, even for float point values. This may be too restrictive. But it can be good to be restrictive at the beginning.
BUG=
Review-Url: https://codereview.webrtc.org/1969403003
Cr-Commit-Position: refs/heads/master@{#12871}
The following algorithmic functionality was added:
-Add support for an exact regressor power to be computed
which avoids the issue with the updating of the filter
sometimes being unstable.
-Lowered the fixed step size of the adaptive filter to 0.05
which significantly reduces the sensitivity of the
adaptive filter to near-end noise, nonlinearities,
doubletalk and the unmodelled echo path tail. It also
reduces the tracking speed of the adaptive filter but the
chosen value proved to give a sufficient tradeoff for the
requirements on the adaptive filter.
To allow the new functionality to be selectively applied the following was done:
-A new Config was added for selectively activating the functionality.
-Functionality was added in the audioprocessing and echocancellationimpl classes
for passing the activation of the functionality down to the AEC algorithms.
To make the code for the introduction of the functionality clean,
the following refactoring was done:
-The selection of the step size was moved to a single place.
-The constant for the step size of the adaptive filter in extended filter mode was
made local.
-The state variable storing the step-size was renamed to a more describing name.
When the new functionality is not activated, the changes
have been tested for bitexactness on Linux.
TBR=minyue@webrtc.org
BUG=webrtc:5778, webrtc:5777
Review URL: https://codereview.webrtc.org/1887003002
Cr-Commit-Position: refs/heads/master@{#12384}
are active in the module and its submodules.
BUG=webrtc:5778, webrtc:5777
Review URL: https://codereview.webrtc.org/1886233003
Cr-Commit-Position: refs/heads/master@{#12371}
where the other audioprocessing unittests are located.
BUG=webrtc:5298
Review URL: https://codereview.webrtc.org/1846323002
Cr-Commit-Position: refs/heads/master@{#12343}
were added in https://codereview.webrtc.org/1773173002.
The reason for the revert is that for some scenarios
that CL causes problems in the coherence estimate used
in the AEC, which in turn causes echo leakage.
The reason for not reverting the actual CL is that
it would cause subsequent CLs to be reverted as well.
Therefore the choice was made to in this CK
instead revert the effects of that CL.
With the changes in this CL, the behavior is bitexact
to what it was before the CL mentioned above.
TBR=aluebs@webrtc.org
BUG=webrtc:5725
Review URL: https://codereview.webrtc.org/1867483003
Cr-Commit-Position: refs/heads/master@{#12259}
Revert reason: I unintentionally added a patch when rebasing that is breaking the bots.
This reverts commit 98c69a0ee785adeb9d95fffeb55cdb6cedbe82c6.
BUG=
Review URL: https://codereview.webrtc.org/1837313002 .
Cr-Commit-Position: refs/heads/master@{#12148}
This CL also extracts part of the functionality used
in the bitexactness test for the high-pass filter into
a separate file in order to be able to reuse that
functionality in bitexactness tests for the other
submodules in APM (including the bitexactness test for
the noise suppressor).
BUG=wertc:5336
Review URL: https://codereview.webrtc.org/1783203002
Cr-Commit-Position: refs/heads/master@{#12061}
It would be good to have a dedicated DebugDumpReplayer. There is one but it hides itself in DebugDumpTest.
This CL is to separate it out.
BUG=
Review URL: https://codereview.webrtc.org/1810463002
Cr-Commit-Position: refs/heads/master@{#12029}
The restriction has been removed completely and AECM now supports any
number of higher bands. But this has been achieved by always zeroing out the
higher bands, instead of applying a constant gain which is the average over half
of the lower band (like it is done for the AEC), because that would be
non-trivial to implement and we don't want to spend too much time on AECM, since
we want to get rid of it in the long term anyway.
R=peah@webrtc.org, solenberg@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1774553002 .
Cr-Commit-Position: refs/heads/master@{#11931}
audio processing module.
The test also adds a new helper class called
VectorBasedAudioFrame that is intended to be
reused for the bitexactness tests for the other
submodules.
BUG=webrtc:1091
Review URL: https://codereview.webrtc.org/1510493004
Cr-Commit-Position: refs/heads/master@{#11864}
The audio level of the AEC's output level was calculated before overlapping add, and therefore, a compensation was needed. The compensation is multiplying the level by 2 since, before overlapping add, the level is roughly halved due to windowing.
This had to be that way because the level was calculated in frequency domain and the signal after overlapping add has only its time domain representation.
The level calculation has been updated to work on time domain signal and therefore the problem is not there any longer.
This CL is to put the calculation of the AEC output level after overlapping add and remove the compensation.
BUG=
R=peah@webrtc.org
Review URL: https://codereview.webrtc.org/1644133002 .
Cr-Commit-Position: refs/heads/master@{#11810}
This is needed when synthesizing a call based on
48 kHz audio files as otherwise an error is
generated about the wrong sample rate is generated.
That error is in turned caused by the sample rate
being changed from the default 16 kHz
at the first Capture API call event.
BUG=
Review URL: https://codereview.webrtc.org/1698243003
Cr-Commit-Position: refs/heads/master@{#11635}
