This CL ensures that the linear echo prediction mode is not used
when the transparent mode is active.
TBR: saza@webrtc.org,gustaf@webrtc.org
Bug: webrtc:9612,chromium:873074
Change-Id: I25cda5226251df769b6524594ea8a2b78532aaec
Reviewed-on: https://webrtc-review.googlesource.com/93740
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24268}
This CL changes the AEC3 code to allow the main and shadow filters
to have different lengths.
Bug: webrtc:9614,chromium:873100
Change-Id: I3ec2861d496986610d5a73db5771bbe9b8bf7dcd
Reviewed-on: https://webrtc-review.googlesource.com/93465
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24265}
This CL makes the jump-starting of the shadow filter more extreme.
It furthermore utilizes this to allow the AEC to rely further, and
more quickly on its linear filter estimates.
The result is mainly increased transparency but also some
cases of fewer echo blips.
Bug: webrtc:9612,chromium:873074
Change-Id: I90f7cfbff9acb9d0c36409593afbf476e7a830d3
Reviewed-on: https://webrtc-review.googlesource.com/93461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24264}
This CL ensures that the shadow filter is adapted at each block, which
avoids that a temporary filter length mismatch can occur between the
main and shadow filters.
Bug: webrtc:9602,chromium:872201
Change-Id: I651812b4e3b134c6c5e1fe3df5ab78dbdb5c1fb4
Reviewed-on: https://webrtc-review.googlesource.com/93000
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24253}
Any modification of the render stream now happens *before* the
echo detector enqueues render stream frames. In this way, there
is no impact of the render pre-processor on the echo likelihood
metric.
Bug: webrtc:9591
Change-Id: I9b5e339e892796a0d0cd072fdd45d35ec89d8802
Reviewed-on: https://webrtc-review.googlesource.com/93031
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24251}
This CL adds a flag to optionally disable the digital gain control in
ExperimentalAgc. With the flag, Experimental Agc (henceforth AGC1)
only controls the adaptive analog gain. This flag can be combined to
that which activates AGC2. That way, one can enable the hybrid AGC
configuration AGC1 analog only + AGC2 fixed+adaptive digital.
Previously, there was a flag "use_agc2_digital_adaptive" in
AgcManagerDirect. Our ambition was that to activate the hybrid mode
described above with this flag. The behavior of the flag was not
implemented.
To activate the hybrid mode after this CL, set
ExperimentalAgc::digital_adaptive_disabled=true and
AudioProcessing::Config::GainController2::enabled=true.
We also add flags for these settings in audioproc_f.
Then the required settings are currently
audioproc_f --agc2 1 --agc 1 --experimental_agc 1 \
--experimental_agc_disable_digital_adaptive 1 \
-i [INPUT]
Bug: webrtc:7494
Change-Id: Iea798dc3899cec83d30ba71caba787262fcaef41
Reviewed-on: https://webrtc-review.googlesource.com/89740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24249}
Agc2 applies a digital gain to the nearend signal.
When the analog level changes, the digital gain calculation is no
longer valid. Therefore Agc2 should be notified to analog gain
changes.
This CL also allow audioproc_f to chain AGC1 and AGC2. In a dependent
CL we will allow using AGC1 for analog gain and AGC2 for digital
gain.
Bug: webrtc:7494
Change-Id: Id75b3728fbf2de1d84b7fba005e4670c7a2985d9
Reviewed-on: https://webrtc-review.googlesource.com/89387
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24231}
This CL adds functionality to use the shadow filter output instead
of the main filter output for cases when the former is better than
the latter. One case when that happens is when there have been an
echo path change, either in the acoustic path, in the audio buffers
or due to some active audio processing effects being applied on
the device.
The CL causes less echo leaks, in particular on devices with
active render processing.
Bug: webrtc:9581,chromium:869821
Change-Id: Icb8df1b94141598da82dc188051ac59e43338938
Reviewed-on: https://webrtc-review.googlesource.com/91820
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24166}
Protect from negative second moments, which are unexpected in TransientDetector::Detect
and may lead to invalid results.
Bug: chromium:866925
Change-Id: Id1d5b2ebb51e54d9d332b869c6f63dcd03cc461c
Reviewed-on: https://webrtc-review.googlesource.com/91164
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24153}
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.
This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org
Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
Related bug closed since half a year back.
Bug: webrtc:8665
Change-Id: I77007caaa97b5db04f5cf144323cac7a576a7fde
Reviewed-on: https://webrtc-review.googlesource.com/90872
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24135}
This CL increases the AEC3 transparency during call
startup and after echo path delay changes in 3 ways:
1. The exit requirements for the initial mode is
made less strict.
2. The requirements for using the linear echo model
are made less strict.
3. The duplicated reverb modelling in the linear mode
removed.
Bug: webrtc:9572,chromium:868329
Change-Id: I79ea0796ed26408e35576bb39eaae4e4848b4f83
Reviewed-on: https://webrtc-review.googlesource.com/90868
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24132}
This will be the one way of toggling AEC. The EchoControlMobile and
EchoCancellation interfaces will be removed.
The settings introduced here are not used yet, to allow for smooth
downstream fixes.
Bug: webrtc:9535
Change-Id: I3b1a524a0ab7daf63419d7e5ed47417b9282dbf6
Reviewed-on: https://webrtc-review.googlesource.com/90864
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24129}
This CL adds a functionality that jump-starts the
AEC3 shadow filter whenever it performs consistently
worse than the main filter.
The jump-start is done such that the shadow filter
is re-initialized using the main filter coefficients.
The effects of this is a significantly more accurate
main linear filter which leads to less echo leakage
and better transparency
Bug: webrtc:9565, chromium:867873
Change-Id: Ie0b23cd536adc7ce96fc3ed2a7db112aec7437f1
Reviewed-on: https://webrtc-review.googlesource.com/90413
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24117}
This CL corrects the way that the echo subtractor output is
adjusted during the adjustment of the adaptive filter when the
analog AGC gain changes.
The CL also ensures that the main adaptive filter is not updated
when this occurs.
Bug: webrtc:9561,chromium:867373
Change-Id: I636f936128f7d9f0d82ca4140b59f148eb35d6a4
Reviewed-on: https://webrtc-review.googlesource.com/90401
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24101}
All clients who do not own their own APM turn it off by default
(in WebrtcVoiceEngine). AECM with comfort noise is a little-exercised
code path. Configurability of this setting is going away, so we're
better off disabling it by default.
Bug: webrtc:9535
Change-Id: Iba839aa18e79ae29ff20bdf6e30de77870ba4143
Reviewed-on: https://webrtc-review.googlesource.com/89583
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24078}
This reverts commit 771b50ca0b.
Reason for revert: Introduces error-prone config.
Original change's description:
> Add one-stop-shop for built-in AEC toggling in APM
>
> This does not change what AEC functionality is available.
> However, a client that only uses this interface - and not the submodule
> pointer accessors - gets simpler code, and is guaranteed not to run any
> two AECs in tandem.
>
> The submodule interface EchoControlMobile is being deprecated in
> https://webrtc-review.googlesource.com/c/src/+/89392
>
> Bug: webrtc:9535
> Change-Id: Id9326074e566be6d8768010fc421c457beff402c
> Reviewed-on: https://webrtc-review.googlesource.com/89386
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24066}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I43283a1b22538a4caa77313499989146b2ce67f1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/90060
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24067}
This does not change what AEC functionality is available.
However, a client that only uses this interface - and not the submodule
pointer accessors - gets simpler code, and is guaranteed not to run any
two AECs in tandem.
The submodule interface EchoControlMobile is being deprecated in
https://webrtc-review.googlesource.com/c/src/+/89392
Bug: webrtc:9535
Change-Id: Id9326074e566be6d8768010fc421c457beff402c
Reviewed-on: https://webrtc-review.googlesource.com/89386
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24066}
In AgcManagerDirect::UpdateGain(), Agc::GetRmsErrorDb() is
called. Depending on the result of that call, the analog gain may be
changed. After an analog gain change, the Agc should be reset, because
it's memory contains now invalid loudness levels.
The Agc in modules/audio_processing/agc/agc.cc resets itself at every
successful Agc::GetRmsErrorDb call. The AdaptiveModeLevelEstimatorAgc
does not. This change makes sure all Agcs are reset from
AgcManagerDirect.
It will cause some Agcs to be reset twice. This is fine, because
Agc::Reset() is cheap and idempotent.
Bug: webrtc:7494
Change-Id: Iee7495d699cbdb9d69b2ae0cb07034c6e2761e22
Reviewed-on: https://webrtc-review.googlesource.com/89040
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24054}
A 32-bit number overflows. It's then capped to compute a 16-bit value.
This CL introduces a 64-bit variable on which equivalent operations are
performed instead.
Bug: chromium:864883
Change-Id: I371af869c6586256b900356491f467bed357e11d
Reviewed-on: https://webrtc-review.googlesource.com/89584
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24041}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251
Change-Id: Ibdafc0bb08de1be7189af7053a67a24e3a26bd6b
Reviewed-on: https://webrtc-review.googlesource.com/89001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23997}
This CL re-activates the explicit handling of microphone
gain changes in the AEC3 code. The implementation is done
beneath a kill-switch so that when that switch is active
the changes in this CL are bitexact.
Bug: webrtc:9526,chromium:863826
Change-Id: I58e93d8bc0bce7bec91e102de9891ad48ebc55d8
Reviewed-on: https://webrtc-review.googlesource.com/88620
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23986}
This CL refactors the code in AEC3 that analyzes how
well the adaptive filter performs. The purpose of this
is both to simplify code that is more complex than needed
and also to pave the wave for the upcoming CLs that
softens the echo suppression during doubletalk.
The main changes are that:
-The shadow adaptive filter is now never analyzed. This
turned out to never affect the output in the recordings
it was tested on.
-The convergence analysis was moved to the aec state
code.
The changes are bitexact on all testcases where they
have been tested on.
Bug: webrtc:8671
Change-Id: If76b669565325c8eb4d11d1178a7e20306da9a26
Reviewed-on: https://webrtc-review.googlesource.com/87430
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23958}
This CL changes a constant from an approximately correct limit
of 2^25.5.
The new limit is the largest x such that z = 10 satisfies:
((x >> z) + 1)^2 <= 2^31 - 1.
If gains[k + 1] > x, then z >= 11 and needs to be computed.
Bug: chromium:860638
Change-Id: If17f257dacd94806e59e4f32b345a5fb15b4e32b
Reviewed-on: https://webrtc-review.googlesource.com/87583
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23908}
This clarifies dependencies and makes it easier to customize builds
for different binaries.
Also adds BUILD files in aec/ and aecm/.
Moves unit tests to their own target, which subjects them to Chromium
Clang style checks.
The CL contains a fix for a thusly induced warning.
Bug: webrtc:9488
Change-Id: I77b680b42a4dccc5f025005e0890f60b4eaf2961
Reviewed-on: https://webrtc-review.googlesource.com/87304
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23887}
This CL does the following:
1. Adds a new AdaptiveModeLevelEstimatorAgc implementation of the Agc
interface. The new implementation differs from webrtc::Agc by
1. using the AGC2 speech level estimator in
GetRmsErrorDb. webrtc::Agc implements its own with help of
webrtc::LoudnessHistogram.
2. Doesn't forget its past at every GetRmsErrorDb call.
2. Makes AgcManagerDirect use AdaptiveModeLevelEstimatorAgc instead of
webrtc::Agc if the use_agc2_level_estimation flag is set.
Bug: webrtc:7494
Change-Id: I8df3f52e322d433eb5ce5297f4236af2f1877b04
Reviewed-on: https://webrtc-review.googlesource.com/86603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23875}
Splits 'modules/audio_processing:audio_processing' target. The files
in modules/audio_processing/agc now are in targets in that folder.
Reason for doing this was to include
modules/audio_processing/agc/agc.h from another target in the
dependent CL https://webrtc-review.googlesource.com/c/src/+/86603
This could help reducing the binary size in the future.
Bug: webrtc:7494
Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887
Reviewed-on: https://webrtc-review.googlesource.com/87422
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23873}
This reverts commit e90879097c.
Reason for revert: breaking downstream projects
Original change's description:
> IWYU: Add <cmath> for fabsf() and powf(), switch to C++ versions
>
> Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
> Changes on the decay estimation") by including the missing header:
>
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
> reverb_decay_(fabsf(config.ep_strength.default_len)),
> ^~~~~
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
> reverb_decay_(fabsf(config.ep_strength.default_len)),
> ^~~~~
> labs
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
> decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
> ^~~~
>
> While here, also switch to the C++ versions of those functions: std::fabs()
> and std::pow() respectively.
>
> Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
>
> Bug: chromium:819294
> Change-Id: Id803243be8dd17eac95c70a88a37ee2fe1505a5a
> Reviewed-on: https://webrtc-review.googlesource.com/87421
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23870}
TBR=gustaf@webrtc.org,alessiob@webrtc.org,raphael.kubo.da.costa@intel.com,devicentepena@webrtc.org
Change-Id: I22423a2d4201183f70ae084e0e21930367824f1c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:819294
Reviewed-on: https://webrtc-review.googlesource.com/87401
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23871}
Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
Changes on the decay estimation") by including the missing header:
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
labs
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
^~~~
While here, also switch to the C++ versions of those functions: std::fabs()
and std::pow() respectively.
Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
Bug: chromium:819294
Change-Id: Id803243be8dd17eac95c70a88a37ee2fe1505a5a
Reviewed-on: https://webrtc-review.googlesource.com/87421
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23870}
Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
Changes on the decay estimation") by including the missing header:
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
labs
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
^~~~
Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
Bug: chromium:819294
Change-Id: If992e5e473b9d4d0c1b3c1006c3816b7c4eee296
Reviewed-on: https://webrtc-review.googlesource.com/87241
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#23863}
This clarifies the dependencies of utility/ a lot (spoiler:
there are very few) and makes it easier to separate the build
targets for aecm and aec2.
Bug: webrtc:9488
Change-Id: If916f86e80c19d1b650d0908fbe8343ea7c47bd7
Reviewed-on: https://webrtc-review.googlesource.com/87141
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23860}
The main filter is adapted at a lower rate which reduces the risk of
diverging during double talk. The change yields notable transparency
improvements.
Bug: webrtc:9497
Change-Id: Ib23b7a4055d313dede535d2b65dc7e023a2db042
Reviewed-on: https://webrtc-review.googlesource.com/87300
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23858}
In this CL we have introduced changes on the estimation of the decay involved in the exponential modeling of the reverberation. Specifically, the instantaneous ERLE has been tracked and used for adapting faster in the regions when the linear filter is performing well. Furthermore, the adaptation is just perform during render activity.
Change-Id: I974fd60e4e1a40a879660efaa24457ed940f77b4
Bug: webrtc:9479
Reviewed-on: https://webrtc-review.googlesource.com/86680
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23836}
This change simplifies the calculation of the suppression gains.
It also contains a new tuning of the suppressor.
The suppressor behavior is tuned by setting echo-to-nearend ratios
for when the suppressor is to be fully transparent and for when to
fully suppress. An echo-to-masker value determines when the signal
is masked by noise. These three values are specified for low and
high frequencies.
Change-Id: I108e83c8f2a35462085a3fabaebcc02fa3103607
Bug: webrtc:9482
Reviewed-on: https://webrtc-review.googlesource.com/86021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23830}
Much like https://bugs.chromium.org/p/chromium/issues/detail?id=855900,
the int32 gain table isn't always small enough for plain multiplication
with an int16.
This appears fixable through regular fixed-point arithmetic (multiply
out[i][n] by integer and fractional part of gain separately), but it's
less readable.
Bug: chromium:858989
Change-Id: Ie5aac25fd0cca4e51858cba69bde06c54a5d31bf
Reviewed-on: https://webrtc-review.googlesource.com/86602
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23815}
1. Adds support for Reset calls in AGC2. The AGC will be reset during
analog gain changes.
2. Allows AdaptiveModeLevelEstimator to return estimates > 0. This can
happen if the signal gain is too high. It's needed for letting the
analog AGC know that the gain is too high.
Bug: webrtc:7494
Change-Id: I38def17c21cc01c36aaea79a2401d8c2f289407b
Reviewed-on: https://webrtc-review.googlesource.com/79360
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23805}
This CL adds two flags to audioproc_f. The flags control
AgcManagerDirect. The flags are
'--experimental_agc_agc2_level_estimator' and
'--experimental_agc_agc2_digital_adaptive'.
After this CL, the flags are be applied to AgcManagerDirect. The flags
have no effect in release-mode. They cause a crash in debug-mode.
In an upcoming CL, '--experimental_agc_agc2_level_estimator 1' will
replace the speech level estimation in ExperimentalAgc with that of
AGC2.
'--experimental_agc_agc2_digital_adaptive 1' will replace the digital
gain selection and application with that of AGC2.
These audioproc_f will activate both new settings:
./out/Target/audioproc_f --agc 1 --experimental_agc 1
--experimental_agc_agc2_digital_adaptive 1
--experimental_agc_agc2_level_estimator 1 --simulate_mic_gain 1
--simulated_mic_kind 2
See also https://webrtc-review.googlesource.com/c/src/+/79360
Bug: webrtc:7494
Change-Id: If0e65893dffdddb312e553787b8cedaf9a334323
Reviewed-on: https://webrtc-review.googlesource.com/86548
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23802}
The AGC submodule of APM changes analog gain. These gain changes are
typically ignored by the test tool audioproc_f.
There is an option of the test tool to take action on the gain
changes. It's the '--simulate_mic_gain' option. The option converts
the analog gain to a digital gain. The digital gain is applied to the
capture stream.
This change adds a new simulated microphone kind. The new microphone
has a gain curve defined by
modules/audio_processing/agc/gain_map_internal.h. That gain curve
defines how AGC1 expects a microphone to behave.
Bug: webrtc:7494
Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780
Reviewed-on: https://webrtc-review.googlesource.com/86128
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23801}
We dump the compression level from AgcManagerDirect.
We use the same names and structure as in
GainControlForExperimentalAgc.
This is to get Apm dump file names to match in the upcoming AGC
changes: https://webrtc-review.googlesource.com/c/src/+/79360
TBR: alessiob@webrtc.org
Bug: webrtc:7494
Change-Id: I1e6260ea48ffc43f709e4b0c97f843ad9c3d1824
Reviewed-on: https://webrtc-review.googlesource.com/86546
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23800}
In this work the performance of the linear filter is
estimated. The estimation aims at capture situations when the linear
filter is largely over-estimating the echo. In those circumstances,
the linear filter is scaled with the purpose of accelerating its
convergence.
Change-Id: I05ea3739d82838a6f08673432da92125c47943e0
Bug: webrtc:9466,chromium:857426
Reviewed-on: https://webrtc-review.googlesource.com/86133
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23789}
This CL removes the constraint that freezes the filter adaptation
whenever the estimated echo or the prediction error is saturated. This
allows for much more rapid filter recovery in cases where the echo path
gain for some reason changes, such as when the analog AGC gain is
adjusted or the loudspeaker volume is changed.
TBR: devicentepena@webrtc.org
Bug: webrtc:9466,chromium:857426
Change-Id: Ic0b3b03f41f12e9a607aaadd2ee91cbaa16cac52
Reviewed-on: https://webrtc-review.googlesource.com/86124
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23775}
This CL changes the behavior when the main filter diverges.
Instead of entering non-linear mode, the AEC continues to operate in
linear mode but estimates the residual echo differently. R2 is S2
scaled by a factor of 10.
Bug: chromium:857018,webrtc:9462
Change-Id: I41212efe164ad319cf38a163cdf9d3ea151e0997
Reviewed-on: https://webrtc-review.googlesource.com/85981
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23772}
The frequency shape of the echo path has been included in the reverberation model.
Bug: webrtc:9454,chromium:856636
Change-Id: Id2bc3096df31e29328936f94fe965ed1883d70f7
Reviewed-on: https://webrtc-review.googlesource.com/85370
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23746}
common_audio/fft4g.c is third party codem that have to be moved into
third_party folder, so to be able to do it at first we have to extract
it into separate target. It is extracted with corresponding header file
and will be moved in futher CL.
Bug: webrtc:8366
Change-Id: I586ca94d4e9242c23163b987fa334dfa705020ed
Reviewed-on: https://webrtc-review.googlesource.com/85372
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23742}
This CL removes the remaining beamformer parts from the APM.
Bug: webrtc:9402
Change-Id: I9ab2795bd2813d17166ed0925125257b82d98a74
Reviewed-on: https://webrtc-review.googlesource.com/83340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23694}
* Move 'VadWithLevel' to AGC2 where it belongs.
* Remove the vectors from VadWithLevel. They were there to make it work
with modules/audio_processing/vad, which we don't need any longer.
* Remove the vector handling from AGC2. It was spread out across
AdaptiveDigitalGainApplier, AdaptiveAGC and their unit tests.
* Hack the RNN VAD into VadWithLevel. The main issue is the resampling.
Bug: webrtc:9076
Change-Id: I13056c985d0ec41269735150caf4aaeb6ff9281e
Reviewed-on: https://webrtc-review.googlesource.com/77364
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23688}
GainControlImpl was inited with two refs to the APM capture lock. As a
result, it could modify member vars without holding the render
lock. The Process and Analyze calls are not affected, because they are
made from audio_processing_impl when APM holds both locks.
Bug: webrtc:9354
Change-Id: I814b69602280921dda9dc45ffcbddb38de4a3394
Reviewed-on: https://webrtc-review.googlesource.com/84182
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23677}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Reduce noise of the nearend spectrum estimation by averaging multiple
frames.
Bug: webrtc:9420,chromium:853699
Change-Id: Iad7e68b1209a369e263b2d892791943e42bfbb3f
Reviewed-on: https://webrtc-review.googlesource.com/83960
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23655}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_processing'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
This removes beamformer references from audioproc_f, some non-beamformer tests, and a few other bits and bobs.
The beamformer is, after this, very decoupled from the remaining APM code.
Bug: webrtc:9402
Change-Id: Iaafc95517013d7a17723ef2329f17b5e09069bc9
Reviewed-on: https://webrtc-review.googlesource.com/83983
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23649}
Removes the usage of an injected/enabled beamformer in APM, and marks
the API parts as deprecated.
Initialization and process calls are removed, and all enabled/disabled
flags are replaced by assuming no beamforming. Additionally, an AGC test
relying on the beamformer as a VAD is removed.
Bug: webrtc:9402
Change-Id: I0d3d0b9773da083ce43c28045db9a77278f59f95
Reviewed-on: https://webrtc-review.googlesource.com/83341
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23643}
The echo detector is currently stored as a unique_ptr, but when injecting an echo detector, a scoped_refptr makes more sense since the ownership will be shared.
Bug: webrtc:8732
Change-Id: I2180014acb84f1cd5c361864a444b7b6574520f5
Reviewed-on: https://webrtc-review.googlesource.com/83325
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23610}
In this work we introduce some changes on how the reverberation model for AEC3 is applied. Currently, the exponential modelling of the tails is applied over the linear echo estimates. That might result in an overestimation of the reverberation tails under certain conditions. In this work, the reverberation model is instead applied over an estimate of the energies at the tails of the linear estimate.
Additionally, the stationary estimator is changed so it does not disable the aec immediately after a burst of activity.
Bug: webrtc:9384,webrtc:9400,chromium:852257
Change-Id: Ia486694ed326cfe231fc688877c0b9b6e2c450ff
Reviewed-on: https://webrtc-review.googlesource.com/82161
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23599}
This CL ensures that the linear-filter based refined delay is chosen to
match the delay that was detected by the delay estimator during the time
it takes for the linear filter to converge.
Bug: webrtc:9371,chromium:850451
Change-Id: Ib9cf532df0577ceca10a260d9d2deba5306f88bb
Reviewed-on: https://webrtc-review.googlesource.com/81682
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23537}
This filter preprocess the time domain representation of the adaptive
linear filter to avoid low-frequency components causing issues in
the filter analysis.
Bug: webrtc:9343, chromium:848231
Change-Id: I40494959f1b76242a7c9f2a2fc85c2ad4af9e164
Reviewed-on: https://webrtc-review.googlesource.com/79142
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23536}
This change contains a new anti-aliasing filter for the delay estimator
for down-sampling factor 4. The new (elliptic) filter has a much wider
main lobe allowing for faster convergence.
Bug: webrtc:9288,chromium:846615
Change-Id: Id109974a59fe6f48c5e0ccc4f4e06c0d94c8bd03
Reviewed-on: https://webrtc-review.googlesource.com/81680
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23534}
Preparing for changing the filters of the decimator by moving the old
filters to the new zero, pole, gain notation.
Bug: webrtc:9288,chromium:846615
Change-Id: I2b01a2555d34617e0bf251c782703753f72cd56f
Reviewed-on: https://webrtc-review.googlesource.com/81189
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23528}
The signal used for delay estimation at downsampling factor 8 is bandpass
filtered and contains less energy than for other downsampling factors.
This CL adjusts the energy threshold used for determining if there is enough
farend activity to update the matched filters in the delay estimator.
Only downsampling factor 8 is affected.
Bug: webrtc:9288,chromium:846615
Change-Id: I6f38f5609a31e7a08e60571ac75ea75c9962e026
Reviewed-on: https://webrtc-review.googlesource.com/80443
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23486}
CascadedBiQuadFilter can run identical filters multiple times. This CL
allows the use of different filters in each step. This enables the use
of more elaborate filters. The filters are defined by zeros, poles and
gains.
The 'old' way of initializing CascadedBiQuadFilter with a transfer
function and number of filters is left intact.
Bug: webrtc:9288,chromium:846615
Change-Id: Ie4a5b98eba044415571cdcac087b20870a0b5d33
Reviewed-on: https://webrtc-review.googlesource.com/80060
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23473}
In this work, we change the behavior of the gain limiter so it also looks at the energy
on farend around the default delay for deciding the suppression gain
that should be applied at the initial portion of the call.
Bug: webrtc:9311,chromium:846724
Change-Id: I0b777cedbbd7fd689e72070f72237296ce120d3c
Reviewed-on: https://webrtc-review.googlesource.com/78960
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23400}
Letting the delay estimator operate at a sampling frequency of 2 kHz
with audio between 0 and 1 kHz makes it sensitive to noisy environments.
This CL bandpass filters the 16 kHz signal before downsampling to 2 kHz
in a way that the downsampled 2 kHz signal contains audio between 1 and
2 kHz. It also sets downsampling factor 8 as default which significantly
reduces computational complexity.
Bug: webrtc:9288,chromium:846615
Change-Id: Iaf67898a1a14326cd61bb7f81c14d3c12a697c8d
Reviewed-on: https://webrtc-review.googlesource.com/78703
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23395}
Code using the macro change to a plain declaration+init of a local
variable.
Also delete includes of <stdint.h> and <stddef.h> from basictypes.h.
Bug: webrtc:6853
Change-Id: I5ffceb449c1bf8f5badb595d5a343a47b0c6deae
Reviewed-on: https://webrtc-review.googlesource.com/78460
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23377}
This CL fixes the rounding of the estimated average call skew. Before it
was rounded down (toward INT_MIN). Now it is rounded to the nearest integer.
This avoids unnecessary fluctuations of the estimated call skew (and
unnecessary resets).
Bug: webrtc:9283,chromium:888042
Change-Id: Id5b3c593f812f5f9fd3dcdafb7e388a6ef1ac153
Reviewed-on: https://webrtc-review.googlesource.com/77684
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23338}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
The ERLE computation was improved by two means:
- The update function was always called and just parts of the internal code reacts to the converged filter flag
- When computing the ERLE, the ratio of energies is now computed using more points and, therefore, a more robust estimation is achieved.
Bug: webrtc:9284
Change-Id: Ie4f871f19cfad1a13741352ddd7b0a27ad6c3fb6
Reviewed-on: https://webrtc-review.googlesource.com/77767
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23329}
This CL applies a high pass filter to the delay estimator signals which
improves the adaptation of the matched filters in noisy environments.
This results in faster delay estimation.
Bug: webrtc:9288
Change-Id: I8ffe5442eab7ac2f10a7ba236b08a0f07ec90645
Reviewed-on: https://webrtc-review.googlesource.com/77725
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23308}
This CL increases the allowed variations in the API call skew limit in
AEC3.
Bug: webrtc:9283,chromium:888042
Change-Id: Ib5e784c6f3dcf1bf3a2cbfe2b1559953db9227a8
Reviewed-on: https://webrtc-review.googlesource.com/77430
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23305}
This CL turns on the previously implemented AEC3 audibility
improvements, which before has been off by default.
Bug: webrtc:9193,chromium:836790
Change-Id: Ibcd057ba5dd002718d62fd83db33d01d9563b8ea
Reviewed-on: https://webrtc-review.googlesource.com/77123
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23265}
Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502
':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.
Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.
NOTRY=true # want this in and 'ios_api_framework' seems stuck.
Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
CustomProcessing is the interface to injectable audio processing
submodules to AudioProcessing. This CL makes it possible to set
runtime settings on the injected render processing component.
Note that the current runtime setting handling happens on the capture
thread. Therefore, we add another SwapQueue to communicate with the
render thread.
Bug: webrtc:9138, webrtc:9262
Change-Id: I665ce2d83a2b35ca8b25cca813d2cef7bd0ba911
Reviewed-on: https://webrtc-review.googlesource.com/76123
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23236}
This CL finalizes the feature extraction part for the RNN VAD adding
a class that combines a high-pass filter, LP residual computation,
pitch estimation and spectral features extraction.
This CL also includes a minor refactoring of the pitch estimation
library.
Bug: webrtc:9076
Change-Id: I918b9e143bc6dd2bf508a891446067258a68a777
Reviewed-on: https://webrtc-review.googlesource.com/75504
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23235}
Mini-change: add 'webrtc' namespace. The template class AudioFrameView
got declared in the global namespace by mistake. (My fault). Now
fixing.
Bug: webrtc:9262.
Change-Id: I6f2b4ab1ccdb223505e7181b8e6f12f5f23b3684
Reviewed-on: https://webrtc-review.googlesource.com/76140
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23215}
This CL defines SpectralFeaturesExtractor which is responsible for
computing the spectral features used as input for the RNN.
Bug: webrtc:9076
Change-Id: I5e1396b89eca9c13bb268e8419a16436a9c3450f
Reviewed-on: https://webrtc-review.googlesource.com/73760
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23206}
This CL adds robustness to avoid the AEC3 transparent mode to be
incorrectly activated when
-there is strong near-end noise
-there is only low-level nearend activity.
Bug: webrtc:9256,chromium:841193
Change-Id: I26c2759d163914eb85dc3d863da8acbf28cbb88d
Reviewed-on: https://webrtc-review.googlesource.com/75511
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23191}
This CL corrects the usage of the estimated echo path gain to not be
hardcoded to 1. In order to retain the tuned behavior, the CL for now
maintains the former behavior in the code.
Bug: webrtc:9255,chromium:851187
Change-Id: I7f91c72e476680a8a854c22b74b1771fae446110
Reviewed-on: https://webrtc-review.googlesource.com/75510
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23190}
This CL ensures that the external audio buffer delay is correctly used
by removing the applied headroom and avoiding that the delay estimation
feedback fromt the echo remover overrules the external delay
information.
Bug: webrtc:9241,chromium:839860
Change-Id: I53cc78ace34a71994ab24a3b552f29979e2aae78
Reviewed-on: https://webrtc-review.googlesource.com/75513
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23189}
During pitch search in the RNN VAD, we calculate auto
correlation. Before this CL, we computed kNumInvertedLags12kHz=147 dot
products of vectors with kBufSize12kHz-kMaxPitch12kHz=240
elements. This was the most time consuming step of the new VAD.
This CL makes the computation happen in frequency domain. Profiling
shows a 3x speed increase. In future, we can try using a more efficient
FFT and to reduce the FFT length to some of e.g. 400, 405, 432.
# For minimal Clang plugin check change.
TBR: kwiberg@webrtc.org
Bug: webrtc:9076
Change-Id: I688251a415869d53175a37f390f441d4e035d954
Reviewed-on: https://webrtc-review.googlesource.com/73366
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23171}
This CL adds helper functions to be used for the spectral features
computation. Namely, it includes the following:
- band boundaries (frequency to FFT coeffcient index)
- band energy coefficients
- log band energy coefficients
- fixed size DCT table and computation
Bug: webrtc:9076
Change-Id: I03a8799b226d986bc1e37cefd0c3039f94b5592a
Reviewed-on: https://webrtc-review.googlesource.com/73687
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23170}
BandAnalysisFft class that wraps the FFT library, makes it easy to change
FFT library, applies windowing function and owns the FFT input buffer.
Bug: webrtc:9076
Change-Id: I9e7ed587ae263b906e04a66bf8c06eaae64daf19
Reviewed-on: https://webrtc-review.googlesource.com/72900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23150}
This CL softens the effect of the AEC3 transparent mode to also handle
headsets that leak low-level echoes in a nonlinear way.
This is handled by reintroducing the limit in the echo path gain for the
nonlinear mode. Due to recent improvements in echo suppressor behavior
this is now possible to do with a limited impact on the near-end speech.
Bug: webrtc:9246,chromium:840347
Change-Id: I0ca5157160d1884ba93b962323b56016756986d3
Reviewed-on: https://webrtc-review.googlesource.com/74703
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23145}
Adding a build target for the bi-qaud filter to make it available for
the RNN VAD of AGC2. Also adding a unit test to test the computation
both in-place and not in-place while comparing the produced output to
that of scipy.signal.
Bug: webrtc:9076
Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772
Reviewed-on: https://webrtc-review.googlesource.com/74220
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23141}
Fixes a confusion of time units (milliseconds vs blocks) of externally
reported audio delay. This fix reduces the risk of echo in the beginning
of a call.
Bug: webrtc:9241,chromium:839860
Change-Id: I534cc15d6b215a5881ae46759f573a56871170a3
Reviewed-on: https://webrtc-review.googlesource.com/74589
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23128}
This CL contains changes to the echo suppressor that improves the
transparency of AEC3.
- The comfort noise level is used as masker and the masking threshold is
increased.
- Suppression gains are allowed to increase more rapidly.
- Suppression gains decrease slower in the lower frequencies after strong
nearend.
Change-Id: I7adf31ed90b0e007072191f40439f27c3b0bccf2
Bug: webrtc:9230,chromium:839379
Reviewed-on: https://webrtc-review.googlesource.com/73680
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23115}
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.
Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
RNN implementation for the AGC2 VAD that includes a fully connected
layer and a gated recurrent unit layer.
Bug: webrtc:9076
Change-Id: Ibb8b0b4e9213f09eb9dbe118bbdc94d7e8e4f91b
Reviewed-on: https://webrtc-review.googlesource.com/72060
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23101}
Functions to estimate pitch period and gain.
Bug: webrtc:9076
Change-Id: Icfe9430dcae11bdb96165c5bfe6e2b1d3bf848ab
Reviewed-on: https://webrtc-review.googlesource.com/70382
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23066}
This CL removes the updating of the buffered data used to to pad the
64 sample blocks to 128 samples FFTs. As that padding was used
incorrectly in one place this resolves an important issue.
Bug: webrtc:9159,chromium:833801,webrtc:9206
Change-Id: Ie6830878ebec6130b61d4e7e3169357f2e253073
Reviewed-on: https://webrtc-review.googlesource.com/73240
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23059}
This CL changes the way the suppressor gain is computed in AEC3 in that
the FFTs used are padded with data and windowed with a Hanning-style
window.
This gives better FFT accuracy, an behavior matching the suppressor
gain application, and also results in one less FFT operation.
Bug: webrtc:9204,chromium:837563
Change-Id: I612676c389cb76a3130966a9b596ff3f44d21863
Reviewed-on: https://webrtc-review.googlesource.com/73141
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23057}
If the adaptive gain is too low, we raise it slowly and only during
speech.
The CL gives better behavior at the start of a call. If the gain is too
high, the fixed-digital limits it. The gain is also quickly reduced by
the AdaptiveGainApplier.
Bug: webrtc:7494
Change-Id: I683f1e3e463cddec2d91f6c7f15c73e744430034
Reviewed-on: https://webrtc-review.googlesource.com/71484
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23053}
This reverts commit b04e5cae08.
Reason for revert: The reason for the revert is that some scenarios were detected where this caused the delay estimation to occur too slowly.
Original change's description:
> Making the delay estimator more robust to noisy nearends and low echoes
>
> This CL reduces the delay estimator step size to make it react better in
> scenarios where the environment is noisy, or the echo level is fairly
> low.
>
> Bug: webrtc:9177,chromium:835281
> Change-Id: I482d898c91eddc497e1284ee500d26df21a0574a
> Reviewed-on: https://webrtc-review.googlesource.com/71486
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22990}
TBR=gustaf@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9177, chromium:835281
Change-Id: I33e09ebfed8ad8330419e554f482c956608befce
Reviewed-on: https://webrtc-review.googlesource.com/72843
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23042}
This CL makes sure that the coherence-based gains are affected by the
upper gain limit during call start-up and after resets.
Bug: webrtc:9159,chromium:833801
Change-Id: I93fdd173b6e11ea861d0e01e12c048ec0a91db70
Reviewed-on: https://webrtc-review.googlesource.com/72841
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23039}
This CL is created from a work initiated at https://webrtc-review.googlesource.com/c/src/+/61160
The purpose of this work is to improve the performance of the echo canceler (AEC3) when the farend signal contains stationary noises:
- An stationarity estimator of the farend signal has been added for detecting the portions of the farend signal that are pure noise.
- When the echo canceler deals with a portion of the signal that contains basically noise, the echo suppressor is able to back-off and avoid the fading of the nearend speech.
Change-Id: Id4b87fc59f4765bf1fca36d1cab39a49aabe104a
Bug: webrtc:9193,chromium:836790
Reviewed-on: https://webrtc-review.googlesource.com/64141
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23024}
The code that attenuates narrow banded echo peaks in low frequencies
is removed as it affects transparency negatively.
Bug: webrtc:9192,chromium:836729
Change-Id: Ib90ce6a3db0a75e8d69bdca432e1f8f8bfbbd988
Reviewed-on: https://webrtc-review.googlesource.com/72380
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23022}
This CL overrides the power-based suppressor gain decision with
a coherence based descision for the cases when that indicates a
higher suppressor gain.
Bug: webrtc:9159,chromium:833801
Change-Id: I0e7d82ac1b8c70ffe9d45907559bb14b1b849d71
Reviewed-on: https://webrtc-review.googlesource.com/71660
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22997}
This CL reduces the delay estimator step size to make it react better in
scenarios where the environment is noisy, or the echo level is fairly
low.
Bug: webrtc:9177,chromium:835281
Change-Id: I482d898c91eddc497e1284ee500d26df21a0574a
Reviewed-on: https://webrtc-review.googlesource.com/71486
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22990}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
- No need to have a unique ptr for the swap queue
- Remove default case from the switch in
AudioProcessingImpl::HandleRuntimeSettings()
Bug: webrtc:9138
Change-Id: I346ba1db6510b5caa637510298b67ead07197b81
Reviewed-on: https://webrtc-review.googlesource.com/71164
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22958}
Functions to estimate the inverse filter via LPC and compute the LP
residual applying the inverse filter.
This CL also includes test utilities, in particular BinaryFileReader,
used to read chunks of data and optionally cast them on the fly, and
Create*Reader() functions to read resource files available at test
time.
Bug: webrtc:9076
Change-Id: Ia4793b8ad6a63cb3089ed11ddad89d1aa0b840f6
Reviewed-on: https://webrtc-review.googlesource.com/70244
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22946}
Adding a data structure to cache the results of pair-wise comparisons
between items stored in a ring buffer. This is used to avoid recomputing
the pair-wise comparison every time that a new item is added in a ring
buffer.
Bug: webrtc:9076
Change-Id: I88fb67a80bd3fd8497764dc7ae7e0a577c06b20f
Reviewed-on: https://webrtc-review.googlesource.com/70162
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22942}
Ring buffer template for a finite number of arrays of given type and size.
Bug: webrtc:9076
Change-Id: Ia6c2065b0013f4a00f693966641f9aebe09f6f5c
Reviewed-on: https://webrtc-review.googlesource.com/70161
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22939}
- protobuf library
- file_wrapper.h
These appear to have been left behind during the AecDump refactoring.
After this CL, APM no longer depends on zlib by default! :)
Bug: webrtc:9139
Change-Id: I12a8df2a17a575515b9c07165825f0879c4e15eb
Reviewed-on: https://webrtc-review.googlesource.com/70762
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22923}
The SequenceBuffer class template implements a linear buffer with a Push
operation that is used to add a fixed size chunk of new samples into the
buffer. Its properties are its size and the size of the chunks that are
pushed. It is used to implement the pitch buffer in the RNN VAD feature
extractor, for which a ring buffer would be a painful choice.
Bug: webrtc:9076
Change-Id: I4767bf06d5a414dbed724a96ea4186ef013a1e30
Reviewed-on: https://webrtc-review.googlesource.com/70204
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22919}
This CL adds support for using any externally reported audio buffer
delay to set the initial alignment in AEC3 which is used before the
AEC has been able to detect the delay.
Bug: chromium:834182,webrtc:9163
Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb
Reviewed-on: https://webrtc-review.googlesource.com/70580
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22917}
This CL increases the transparency in AEC3 during regions of low level
echo. What is done is:
-Low-level echoes are smoothly weighted so as to be deemed less
disturbing.
-The time-domain masking effect of the nearend speech is increased for
all frequencies.
-A separate, even more increased, time-domain masking effect is
introduced for lower frequencies.
-The intra-band masking is reduced to reduce the risk of echo leakage.
-The limiting of maximum gain due to filter-bank dynamics is removed
as the usecase for it could no longer be identified.
Bug: webrtc:9159,cromium:833801
Change-Id: I289b92919763124d6c5e5ede19e9a5917877c654
Reviewed-on: https://webrtc-review.googlesource.com/70421
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22915}
This reverts commit 8628f5bb7c.
Reason for revert: iOS buildbot failing
Original change's description:
> AGC2 RNN VAD: initial build targets
>
> rnn_vad_tool is an executable that reads a wav file of any sample rate
> compatible with 10 ms frames that are resampled and, when the VAD is
> fully landed, will process the resampled frames to compute the VAD
> probability.
>
> To avoid mac, win and ios trybot failures, to_be_removed.h/.cc have
> been added and will be removed as soon as the :lib target includes
> code that leads to a non-empty static lib file on those platforms.
>
> Bug: webrtc:9076
> Change-Id: I810c08acfa1adf2029e3baac2adda3045ae5214a
> Reviewed-on: https://webrtc-review.googlesource.com/70202
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22898}
TBR=alessiob@webrtc.org,aleloi@webrtc.org
Change-Id: Ic6014dde78b0ef371804c52608145ba8acdd9c97
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9076
Reviewed-on: https://webrtc-review.googlesource.com/70144
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22899}
rnn_vad_tool is an executable that reads a wav file of any sample rate
compatible with 10 ms frames that are resampled and, when the VAD is
fully landed, will process the resampled frames to compute the VAD
probability.
To avoid mac, win and ios trybot failures, to_be_removed.h/.cc have
been added and will be removed as soon as the :lib target includes
code that leads to a non-empty static lib file on those platforms.
Bug: webrtc:9076
Change-Id: I810c08acfa1adf2029e3baac2adda3045ae5214a
Reviewed-on: https://webrtc-review.googlesource.com/70202
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22898}
Commit bbf21a3fd6 ("Remove dependencies on
modules:module_api from AudioProcessing") causes the build to fail with
libstdc++ due to several files using memcpy(3) or memset(3) while relying on
string.h being included implicitly by other headers.
Bug: webrtc:9139
Change-Id: Ib73284962f8694d8bed0551968265bfd13cab967
Reviewed-on: https://webrtc-review.googlesource.com/70180
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#22895}
Since we always pass in the first audio channel, we should always pass 1 as the number of channels in the initialization function.
Bug: webrtc:8732
Change-Id: I978edb125d7cc701a5e07193256327908be00560
Reviewed-on: https://webrtc-review.googlesource.com/69660
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22885}
It's a module for applying a gain to the capture signal.
The gain is the first processing step in APM.
After this CL, these two features work:
* The PreAmplifier can be activated with
AudioProcessing::Config::pre_amplifier
* The PreApmlifier can be controlled after APM creation by
AudioProcessing::SetRuntimeSetting.
What's left is a change to AecDumps and to AecDump-replay.
NOTRY=True # 1-line change, tests just passed.
Bug: webrtc:9138
Change-Id: I85b3af511695b0a9cec2eed6fee7f05080305e1d
Reviewed-on: https://webrtc-review.googlesource.com/69811
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22881}
Add configuration fields for the pre-amplifier in the Audio Processing
Module. Also add flags and settings for the pre-amplifier in
audioproc_f.
Also make the setting stored in Aec Dumps. And make the setting
applied when playing back Aec Dumps in audioproc_f.
Bug: webrtc:9138
Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3
Reviewed-on: https://webrtc-review.googlesource.com/69560
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22876}
This CL includes the following changes:
- APM runtime setting (ID + float payload) and unit tests
- Swap queue of APM runtime settings used in AudioProcessingImpl
- runtime settings handler that forwards the settings to APM
sub-modules when a message is retrieved from the queue
- Unit test placeholder to check that the pre-gain update message
is correctly delivered
Bug: webrtc:9138
Change-Id: Id22704af15fde2b87a4431f5ce64ad1aeafc5280
Reviewed-on: https://webrtc-review.googlesource.com/69320
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22873}
Added a new sub-module 'GainApplier'. The build target is
'modules/audio_processing/agc2:gain_applier'. A small refactoring
makes the GainApplier used in adaptive-digital AGC2.
The AGC2 now multiplies samples with a gain in 3 places. It's the
GainApplier, the GainCurveApplier, and the FixedGainController. The
GainApplier is used in AdaptiveDigitalGainApplier and will be used as
a pre-amplifier.
Bug: webrtc:9138
Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6
Reviewed-on: https://webrtc-review.googlesource.com/69321
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22849}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This CL changes the handling of saturated microphone signals in AEC3.
Some of the changes included are
-Make the detection of saturated echoes depend on the echo path gain
estimate.
-Remove redundant code related to echo saturation.
-Correct the computation of residual echoes when the echo is saturated.
-Soften the echo removal during echo saturation.
Bug: webrtc:9119
Change-Id: I5cb11cd449de552ab670beeb24ed8112f8beb734
Reviewed-on: https://webrtc-review.googlesource.com/67220
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22809}
instead of relying on optional.h to included these 2 headers.
Bug: webrtc:9078
Change-Id: I7a4b3facd81690b8f107640487e129986c1f5ff6
Reviewed-on: https://webrtc-review.googlesource.com/68602
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22803}
|is_posix| will be switched to false for Fuchsia, this is a preliminary change.
Bug: chromium:812974
Change-Id: I3bfda3e056ad1e5229834286ce5d095d9204a428
Reviewed-on: https://webrtc-review.googlesource.com/65782
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Fabrice de Gans-Riberi <fdegans@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22753}
AGC2 component that computes and applies the digital gain.
The gain is computed from an estimated speech and noise level.
This component decides how fast the gain can change and what it
should be.
Bug: webrtc:7494
Change-Id: If55b6e5c765f958e433730cd9e3b2b93c14a7910
Reviewed-on: https://webrtc-review.googlesource.com/64985
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22741}
We put back the old noise estimator from LevelController. We add a few
new unit tests. We also re-arrange the code so that it fits with how
it is used in AGC2. The differences are:
1. The NoiseLevelEstimator is now fully self-contained.
2. The NoiseLevelEstimator is responsible for calling SignalClassifier
and computing the signal energy. Previously the signal type and
energy were used in several places. It made sense to compute the
values independently of the noise calculation.
3. Re-initialization doesn't have to be done by the caller.
4. The interface is AudioFrameView instead of AudioBuffer.
# Bots are green, nothing should break internal stuff
NOTRY=True
Bug: webrtc:7494
Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d
Reviewed-on: https://webrtc-review.googlesource.com/66380
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22738}
Another submodule of the Automatic Gain Controller 2. It refines the
biased estimate of the Adaptive Mode Level Estimator. It works by
generating a delayed stream of peak levels. The delayed peaks are
compared to the level estimate.
Bug: webrtc:7494
Change-Id: If4c2c19088d1ca73fb93511dad4e1c8ccabcaf03
Reviewed-on: https://webrtc-review.googlesource.com/65461
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22732}
We update the configuration settings for AGC2. We also update their
effects. Now, 'gain_controller2.enable=true' means 'first run Adaptive
AGC2; then run AGC2 limiter'.
Previously, only the AGC2 limiter was implemented. To run that, one
had to set both 'gain_controller2.enable=true' and
'gain_controller2.enable_limiter=true'.
This setting also enables adaptive AGC2 in the test tool 'audioproc_f'.
Bug: webrtc:7494
Change-Id: I0d5dfe443f2cdc0ecf3aa4054442dab6276d284d
Reviewed-on: https://webrtc-review.googlesource.com/64990
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22669}
This CL adds a timeout for the detection of the headset mode that
allows it to be entered also for the cases where a headset is
inserted during the call.
Bug: chromium:826720,webrtc:9083
Change-Id: Ic3cb4cc0258997a74eccd1bcdf65765e44016ad8
Reviewed-on: https://webrtc-review.googlesource.com/65240
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22658}
The level estimator (AdaptiveModeLevelEstimator) produces a biased
estimate of the speech level. In our model, we use another module
(the SaturationProtector) to compute the bias. This CL contains the
estimator and a stub of the saturation protector.
Bug: webrtc:7494
Change-Id: I0df736d0346063f544fa680b4cc84177ea548545
Reviewed-on: https://webrtc-review.googlesource.com/64820
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22641}
This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.
Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
1. Level Estimator - it gets the energy and a speech probability
and updates a speech level estimate.
2. Noise Estimator - it gets an immutable view of the speech frame
and updates the noise level estimate
3. Gain applier - it gets the speech frame, the current speech and
noise estimates, and the speech probability. It finds a gain to
apply and applies it.
4. AdaptiveAgc - sets up and controls the sub-modules described
above.
Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
We had the following pattern:
if (case_A) metric = METRIC_A;
if (case_B) metric = METRIC_B;
RTC_HISTOGRAM_COUNTS_10000(metric, value);
That's wrong, because once the logging macro runs once, it will use
the same histogram no matter what the first argument is. The macro
expands into roughly
static Histogram* histogram_ptr = nullptr;
if (histogram_ptr == nullptr) {
// Look up the histogram and put in histogram_ptr
}
// Add data through the histogram pointer.
We change the logging to use macros with string literals. We add a
macro for every of the 4 possible invocations. The macros will expand
to one static pointer each.
Bug: webrtc:8925
Change-Id: Ic7e4a6299eff31dd5988047edfcedce7d369e5ce
Reviewed-on: https://webrtc-review.googlesource.com/64724
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22606}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
The estimation on how well the linear filter in the AEC3 is performing
is done through an estimation of the ERLE. That estimation is then
used for knowing how much the suppressor needs to react in order to
cancel all the echoes.
In the current code, the ERLE is quite conservative during farend
inactivity and it is common that it goes to a minimum value during
those periods. Under highly varying conditions, that is probably the
right approach. However, in other scenarios where conditions does not
change that fast there is a loss in transparency that could be avoided
by means of a different ERLE estimation.
In the current CL, the ERLE estimation has been changed in the
following way:
- During farend activity the ERLE is estimated through a 1st order AR
smoother. This smoother goes faster toward lower ERLE values than to
larger ones in order to avoid overestimation of this
value. Furthermore, during the beginning of the farend burst, an
estimation of the ERLE is done that aim to represent the performance
of the linear filter during onsets. Under highly variant environments,
those quantities, the ERLE during onsets and the one computed during
the whole farend duration, would differ a lot. If the environment is
more stationary, those quantities would be much more similar.
- During nearend activity the ERLE estimation is decreased toward a
value of the ERLE during onsets.
Bug: webrtc:9040
Change-Id: Ieab86370a4333d2d0cd7041047d29651de4f6827
Reviewed-on: https://webrtc-review.googlesource.com/62342
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22568}
This CL adds robustifications for avoiding that the headset mode
is triggered for reverberant or weak echo paths.
Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ib111e617f765377c021a5b633cf13a7917fe62a6
Reviewed-on: https://webrtc-review.googlesource.com/64002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22557}
This CL robustifies the echo removal behavior when headsets are used.
In particular it:
-Introduces a secondary, more refined alignment when no alignment can
be found using the delay estimator.
-Changes decision logic for when to use the linear filter output.
-Changes the decision logic for when to be transparent.
-Changes the way that the transparent mode works.
-Makes the nonlinear mode less aggressive.
-Removes the detector for non-audible echoes.
-Makes the attenuation when there are signals with strong narrowband
characteristics more mild in scenarios with low render.
Furthermore the CL:
-Removes the input of external echo leakage information.
Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ied1fe0c0a35d3c31b47606ed2db319a73644d406
Reviewed-on: https://webrtc-review.googlesource.com/60866
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22548}
And fix typo in UMA metric.
We have this pattern in the FrameCombiner component of the AudioMixer:
if (number_of_streams <= 1) {
// Copy or fill with zeros.
return;
}
// Mix and limit
LogMixingStats(/* args */);
When there is only one remote stream, info about active streams and
sample rate is not logged. This CL moves the call to log stats before
the 'return'.
Bug: webrtc:8925
Change-Id: I7b54f61f628273631909dafbfafa21e155e18d4a
Reviewed-on: https://webrtc-review.googlesource.com/62860
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22493}
This CL significantly improves the response time
of the AEC3 delay estimator to audio buffer issues.
The CL adds ensures that the delay estimator
correlators reacts to buffer issues from the
zero state which is much faster than if it has already
achieved a state matching a previous alignment.
The CL has been extensively tested on offline
recordings.
Bug: webrtc:9023, chromium:822245
Change-Id: Ic149b9429e592d4c3535eb8432582f435a1b4745
Reviewed-on: https://webrtc-review.googlesource.com/62081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22461}
This CL ensures a smooth transition from the parameters used during
the startup phase in the call to the parameters used in the rest of the
call. This is achieved by slowly transitioning between the parameter
sets via interpolation.
Bug: chromium:819240,webrtc:8983
Change-Id: Ifbac4b93fc6ad6efc441f41fb88ef09e8ee3d669
Reviewed-on: https://webrtc-review.googlesource.com/60360
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22451}
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.
Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
We want to know how the AudioMixer is used and how FixedGainController
behaves.
The WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.* metrics measures
how often the input level hits different regions of the Fixed Gain
Controller gain curve (when the limiter is enabled). They also measure
how long the metrics stay in different regions. They are related to
WebRTC.Audio.ApmCaptureOutputLevelPeakRms, but the new metrics measure
the level before any processing done in APM.
The AudioMixer mixes incoming audio streams. Their number should be
mostly constant, and often some of them could be muted. The metrics
WebRTC.Audio.AudioMixer.NumIncomingStreams,
WebRTC.Audio.AudioMixer.NumIncomingActiveStreams log the number of
incoming stream and how many are not muted. We currently don't have
any stats related to that.
The metric WebRTC.Audio.AudioMixer.MixingRate logs the rate selected
for mixing. The rate can sometimes be inferred from
WebRTC.Audio.Encoder.CodecType. But that metric measures encoding and
not decoding, and codecs don't always map to rates.
See also accompanying Chromium CL
https://chromium-review.googlesource.com/c/chromium/src/+/939473
Bug: webrtc:8925
Change-Id: Ib1405877fc1b39e5d2f0ceccba04434813f20b0d
Reviewed-on: https://webrtc-review.googlesource.com/57740
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22443}
Since the echo detector processes both the render and the capture audio streams, it needs to know the sample rates and number of channels of both.
Bug: webrtc:8732
Change-Id: Icd26e561d5dd98bd789a6dfa75f468f3fde06fee
Reviewed-on: https://webrtc-review.googlesource.com/61861
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22436}
This CL corrects the bug that only looked at narrowband
render signals above 900 Hz and only assumed that the
influence of such lasted for 6 blocks, which resulted
in filter divergence and echo leakage.
Bug: webrtc:9008,chromium:821670
Change-Id: I9b2635d24b260e9d9a8c5c088ab663e03fb93c42
Reviewed-on: https://webrtc-review.googlesource.com/61800
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22434}
This CL corrects some errors that were included in the CL for reading
the AEC3 options in the audioproc_f tool
Bug: webrtc:8671
Change-Id: Iecaee0ebf08f8a8f75aba1d395dd467a41b876f3
Reviewed-on: https://webrtc-review.googlesource.com/60870
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22384}
This refactoring makes it easier to experiment with injectable components.
Bug: webrtc:8732
Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217
Reviewed-on: https://webrtc-review.googlesource.com/60863
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22372}
This is a reland of 6f37ed78d9
CQ dry run OK except for missing iOS swarming bots.
NOTRY=True
Original change's description:
> Deprecate the adaptive level controller
>
> Level control handled by default-on AGC.
>
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}
Bug: none
Change-Id: I0b9b8e2f3457d5efd4603efbfbbc6b84651df315
Reviewed-on: https://webrtc-review.googlesource.com/60720
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22352}
This CL extends the options for the audioproc_f tool to match the options
for AEC3.
Bug: webrtc:8671
Change-Id: I39972eae33dba461b94118ec47a8560eb9cfe5a6
Reviewed-on: https://webrtc-review.googlesource.com/43120
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22344}
So that we can avoid dependency cycles.
Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
This CL introduces a different rampup behavir for the call startup and after resets
that may occur due to delay changes, clock-drift and audio path glitches.
Bug: chromium:819111, webrtc:8979
Change-Id: Ied1d7896be7f0c69aa6deb61475117021ca6ab09
Reviewed-on: https://webrtc-review.googlesource.com/60002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22312}
This provides the empty shell of an AudioGenerator class.
It is intended to be used for debugging purposes and can be inserted
into the APM much like an AecDump. It allows for playing out diagnostic
audio unaffected by codecs and network jitter, while still capturing
API interaction like in a normal call.
NOTRY=True
Bug: webrtc:8882
Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7
Reviewed-on: https://webrtc-review.googlesource.com/53300
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22282}
Note: estimation is turned OFF if config_.ep_strength.default_len
is set >= 0 (in this case config_.ep_strength.default_len defines a
constant echo decay factor), and hence turned ON if < 0. In case the
echo tail estimation is turned ON, -config_.ep_strength.default_len is
the starting point for the estimator.
The estimation is done in two passes; first we go through all "sections"
(corresponding to chunks of length kFftLengthBy2) of the filter impulse
response to determine which sections correspond to a "stable" decay",
and then the second pass we go through each stable decay section and
estimate the decay. The actual decay estimation is based on linear
regression of the log magnitude of the squared impulse response.
A bunch of sanity checks are also performed continuously to avoid
estimation error during e.g., filter adaptation.
Bug: webrtc:8924
Change-Id: I686ce3f3e8b6b472348f8d6e01fb44c31e25145d
Reviewed-on: https://webrtc-review.googlesource.com/48440
Commit-Queue: Christian Schuldt <cschuldt@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22247}
The webrtc::AudioMixer uses a limiter component. This CL allows
changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome
field trial.
The AGC2 limiter has a float interface. We plan to eventually switch
to the AGC2 limiter. Therefore, we will now mix in de-interleaved
floats. Float mixing will happen both when using the old limiter and
when using the new one.
After this CL the mixer will support two limiters. The limiters have
different interfaces and need different processing steps. Because of
that, we make (rather big) changes to the control flow in
FrameCombiner. For a short while, we will mix in deinterleaved floats
when using any limiter.
Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/
Reverted in https://webrtc-review.googlesource.com/c/src/+/57940
because of both breaking compilation and having a severe error. The
error is fixed and a test is added. The compilation issue is fixed.
Bug: webrtc:8925
Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0
Reviewed-on: https://webrtc-review.googlesource.com/58085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22207}
The AEC3 factory is now part of the WebRTC API.
Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
The unit tests for AudioSendStream was generating a lot of warnings
about "Uninteresting mock function call" on mocked objects. This is due
to the default gmock implementation being NaggyMock and there was no
NiceMock override.
With this change the mocks are replaced with NiceMock implementations
which do not output warnings for unexpected calls. This makes the error
output from the test runner much easier to visually parse to find the
actual errors in failing tests.
Bug: None
Change-Id: Ic40db78159536ddeaa72a468fc2cb3ec17386d44
Reviewed-on: https://webrtc-review.googlesource.com/56220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22152}
This CL moves the AEC3 API call skew estimator into a separate file.
This has the advantage that it can more easily be tested.
The CL also simplifies the code and adds unittests.
Bug: webrtc:8671
Change-Id: I19bc31ca5666cdc87a1ed14770ef20ead1b5b80d
Reviewed-on: https://webrtc-review.googlesource.com/55860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22144}
This CL adds functionality for passing the information about the
estimated delay to the echo remover in AEC3.
The CL also adds information about how long ago the delay changed,
and how long ago the delay estimate was updated.
Bug: webrtc:8671
Change-Id: If274ffe0465eb550f3e186d0599c6dc6fef7f5e8
Reviewed-on: https://webrtc-review.googlesource.com/55261
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22137}
This CL adds functionality to allow removal of any echo occurring
before the render and capture signals have been properly aligned.
The functionality is added in such a manner that the transparency
to nearend is maintained as much as possible.
Bug: webrtc:8883
Change-Id: I813cbbc4c48822e7dffcd9ab6233be4c222089de
Reviewed-on: https://webrtc-review.googlesource.com/49941
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22107}
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.
The other IGC and LE submodules were added in previous CLs [1] and
[2].
This CL also turns on AGC2 in the APM fuzzer.
[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381
Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
Targets containing files in api/audio are moved from api/BUILD.gn to
api/audio/BUILD.gn.
Bug: webrtc:8844
Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981
Reviewed-on: https://webrtc-review.googlesource.com/54360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22074}
This CL adds the Level Estimator of the new gain controller. The Level
Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub
frames. We take the maximal sample values in every sub frame. We then
apply attack/decay smoothing. This is the final level estimate.
The results will be used with InterpolatedGainCurve (see this CL
https://webrtc-review.googlesource.com/c/src/+/51920). For every level
estimate value, we look up a gain with
InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to
the signal.
Bug: webrtc:7949
Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1
Reviewed-on: https://webrtc-review.googlesource.com/52381
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22054}
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.
The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().
This CL contains
* build changes to make modules/audio_processing/agc2 an independent
target
* a new MutableFloatAudioFrame which is the audio interface between
AGC2 and APM
* move of the fixed gain application from GainController2 to
FixedGainController.
If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#
Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
The functions replace some existing code and will be used in the
the new AutomaticGainController.
Bug: webrtc:7949
Change-Id: I9a32132d4a4699a507b8548a2eac10972a2f3fd6
Reviewed-on: https://webrtc-review.googlesource.com/53141
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22045}
Avoid including audio_processing.h from within AEC3.
Bug: webrtc:8844
Change-Id: I02c475c2fb84e2c24eac86baac3c7edaa08bebc0
Reviewed-on: https://webrtc-review.googlesource.com/53065
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22029}
This is one of several small steps of separating APM and AEC3.
Bug: webrtc:8844
Change-Id: Ib6e518fec5f7566cab3823ab35fcede8433f8f4e
Reviewed-on: https://webrtc-review.googlesource.com/53142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22028}
This CL resets the AEC3 realignment functionality when a significant
and persistent skew in the number of render and capture API calls is
detected.
Bug: chromium:811658,webrtc:8879
Change-Id: Ib5c727b38f427da2a7d25eac7c939a17bdaabe74
Reviewed-on: https://webrtc-review.googlesource.com/52260
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21997}
This CL adds robustness in terms of echo removal and faster recovery
in order to regain echo canceller transparency after echo path changes.
The CL does:
-Improve the adaptation rate of the linear filter.
-Increase the look-window used before the linear filter has adapted.
-Decrease the effects of missed detection of residual echo.
-Increase the safety margin before allowing the suppressor gain to
increase.
Bug: chromium:804873,webrtc:8788
Change-Id: I28eedc4c8d0a4f0bc7b79c02d6d59bf00fddd566
Reviewed-on: https://webrtc-review.googlesource.com/48721
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21917}
WHAT: made a BUILD.gn with library and tests in the Audio Processing
Module Voice Activity Detector directory. Updated depending
code. Fixed a Clang warning.
WHY: to make it possible for a target to depend on just the VAD and
not the whole APM. There are other benefits:
* Sometimes faster compilation.
* The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making
a peerconnection shared object file without the VAD has to be done in
steps. The first step is a custom target for the VAD. Hence this Cl.
Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af
BUG: webrtc:5716, webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/47521
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21893}
Due to the growing number of arguments, these functions are being replaced by the AudioProcessingBuilder class.
Bug: webrtc:8668
Change-Id: Ic3936fbd47d92eac22a857a678dca5fd8c029d8b
Reviewed-on: https://webrtc-review.googlesource.com/46241
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21826}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
The AecMobile struct contains a ::farendOld field. It's type is 'short [2][80]'.
The field was initialized by
memset(&aecm->farendOld[0][0], 0, 160);
But sizeof(short) is not guaranteed to be 1. This causes use of
unititialized memory on some platforms. According to MSAN, it can
affect the output of the echo canceller.
The issue was found by the MSAN fuzzer.
This change initializes the array properly.
Bug: chromium:805396
Change-Id: Ibcaca2185cfa153e8fd826e9addfc04d7b65e417
Reviewed-on: https://webrtc-review.googlesource.com/43860
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21764}
A coherence vector cohxd is computed in
WebRtcAec_ComputeCoherence. The coherence values should theoretically
be 0 <= x <= 1. Due to the way they are computed that is not always
the case.
The coherence values are used to update an error signal
estimate hNl in webrtc::EchoSuppression. 'hNl[i]' should contain an
error magnitude for frequency 'i'.
The error magnitudes are used as a basis for exponentiation. If a
magnitude is negative, the result is NaN.
The NaNs will then spread to the output signal.
This change caps the hNl values at 0. I considered capping the
coherence values at 1. The coherence values are calculated differently
for MIPS, NEON and SSE. Therefore it's simpler to cap the hNl values
instead.
The issue was found by the AudioProcessing fuzzer.
Bug: chromium:804634
Change-Id: I8ebaa441d77c3f79d9c194a850cb2b9eed1c2024
Reviewed-on: https://webrtc-review.googlesource.com/43740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21761}
This change
* replaces a left shift with multiplication, because the shiftee can
be negative.
* replaces a right shift (a >> b) with the expression (b >= 32 ? 0 : a >> b)
because a is a 32-bit value, and b can be >= 32.
cppreference quote relating to the second change:
"In any case, if the value of the right operand is
negative or is greater or equal to the number of bits in the promoted
left operand, the behavior is undefined."
Bug: chromium:805832 chromium:803078
Change-Id: I67db0c3fedb0af197b2205d424414a84f8fde474
Reviewed-on: https://webrtc-review.googlesource.com/43761
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21760}
The faster AEC3 alignment introduced recently may in
cases cause the alignment (and the AEC3) to repeatedly
reset. This CL avoids these resets by handling buffer
issues (which are triggering the resets) separately
during the initial coarse alignment phase.
Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53
Bug: webrtc:8798,chromium:805815
Reviewed-on: https://webrtc-review.googlesource.com/43480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21758}
This change handles a special case in NoiseSuppression. The special
case was found by the AudioProcessing fuzzer.
A const copy of the capture audio stream is sent to
NoiseSuppression::AnalyzeCaptureAudio. Then audio undergoes processing
by e.g. the echo canceller. Then it's processed by
NoiseSuppression::ProcessCaptureAudio.
The special case is when the following conditions are all satisfied:
* All stream samples are constantly zero in the call to
AnalyzeCaptureAudio
* a processing component modifies it to be nonzero before the call to
ProcessCaptureAudio
* The array NoiseSuppressionC::magnPrevAnalyze is filled with
zeros. This holds after initialization.
In this case, there is a division by zero in WebRtcNs_ProcessCore. The
resulting NaN values pollute the output signal. They are only detected
several submodules later in the process chain. The NaN values cause
the EchoDetector to crash in debug mode.
There is special handling of the case when the signal is constant zero
in ProcessCore. This change avoids zero division by handling this
issue the same way.
Bug: chromium:803810 chromium:804634
Change-Id: I6d698dd0cd27e6d550b42085124300ce58533125
Reviewed-on: https://webrtc-review.googlesource.com/41282
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21745}
This CL increases the speech of the initial alignment in AEC3 by
loosening the requirements on the accuracy of the initial estimates.
Bug: webrtc:8784, chromium:804270
Change-Id: I86e2d97830843524090a1cf877965739f66dc058
Reviewed-on: https://webrtc-review.googlesource.com/40660
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21728}
The conversational_speech_generator tool now adjusts the level of
different speech segments.
Implementation:
The Turn and MultiEndCall::SpeakingTurn structs have an extra 'gain'
member. It's read and parsed in timing.cc and put in a Turn
struct. It's put in a SpeakingTurn struct in multiend_call.cc and read
and applied to the signal in simulator.cc
Bug: webrtc:7494
Change-Id: I9b82a896eb616c8b5ef14d41dfdfd085ef1d3fbb
Reviewed-on: https://webrtc-review.googlesource.com/26280
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21714}
This minor issue was found by the UBSAN fuzzer.
We have used the Godbolt compiler explorer to check that similar
changes produce identical compiled code.
Bug: chromium:803078
Change-Id: Ib3fa38c101d7bda53d8d39062cb2c0a55144305f
Reviewed-on: https://webrtc-review.googlesource.com/42580
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21713}
