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}
