Implement AnalogGainStatsReporter and add it in AudioProcessingImpl.
This class computes statistics for analog gain updates and
periodically reports them into a histogram.
The added histograms for analog gain update statistics:
- WebRTC.Audio.ApmAnalogGainDecreaseRate
- WebRTC.Audio.ApmAnalogGainIncreaseRate
- WebRTC.Audio.ApmAnalogGainUpdateRate
- WebRTC.Audio.ApmAnalogGainDecreaseAverage
- WebRTC.Audio.ApmAnalogGainIncreaseAverage
- WebRTC.Audio.ApmAnalogGainUpdateAverage
The histograms are defined in
https://chromium-review.googlesource.com/c/chromium/src/+/3207987
Bug: webrtc:12774
Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35301}
Remove analog_level_minimum and analog_level_maximum from
AudioProcessing GainController1 and replace their use with fixed
values 0 and 255, respectively.
Bug: webrtc:12774
Change-Id: Ia4bfe5ed43a65f1587ed67f36bfbb2966b6fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235822
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35297}
The class has also been renamed to better reflect its purpose.
Bug: webrtc:7494
Change-Id: I223a364ab4f8b8a5fef765848bf05675d045cefd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236343
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35277}
This CL includes two changes that break bit-exactness, but that haven't
changed the way AGC2 behaves - the new behavior has been verified with
audioproc_f on a collection of AEC dumps and Wav files (42 recordings
in total).
1) The fixed digital controller can directly be initialized in the
`GainController2` ctor. Before, `SetGainFactor()` was called after the
creation of the object and that caused an initial ramp up lasting one
10 ms frame from -inf to 0 dB. As an effect of the new initialization,
the initial ramp up doesn't happen anymore.
2) In [1] the AGC2 VAD has been moved from the adaptive digital
controller into `GainController2`. In order to not break bit-exactness,
the VAD was placed after the fixed digital controller and before the
adaptive digital one. However, to reduce the chance of incorrect
estimation of the speech probability, the VAD should analyze the
audio before any digital processing is applied inside AGC2.
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I9418229cbe537014fed8271c5550c3ce2bc88e26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235240
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35252}
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Id9849c4463791f5a203afe31efc163efb4d4458e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234583
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35248}
Only used in unit tests and a duplication of what `capture_output_rms_`
already does.
This CL also removes `AudioProcessingStats::output_rms_dbfs`, which is
now unused.
Bug: webrtc:5298
Fix: chromium:1261339
Change-Id: I6e583c11d4abb58444c440509a8495a7f5ebc589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235664
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35246}
Also stop using ApplyConfig() and in [1] fix the build errors when
WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined.
[1] modules/audio_processing/test/audio_processing_builder_for_testing.cc
Bug: webrtc:5298
Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35228}
Not passing the sample rate to the `VoiceActivityDetectorWrapper` ctor
yet since that would require an unnecessary refactoring of `AdaptiveAgc`
which will soon be removed.
Instead, to ensure correct initialization until the child CL [1] lands,
`VoiceActivityDetectorWrapper::initialized_` is temporarily added.
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I4b4be7b8106ba36c958d91bf263a7b30271a1ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234587
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35213}
Internal refactoring of AGC2 to decouple the VAD, its wrapper and the
peak and RMS level measurements.
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Ib560f1fcaa601557f4f30e47025c69e91b1b62e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234524
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35208}
When `AudioProcessingImpl::ApplyConfig()` is called, AGC2 is initialized
and then the new config is applied. That is error prone and for example
breaks bit exactness in [1].
Changes:
- `GainController2` must be created by passing configuration,
sample rate and number of channels
- `GainController2::ApplyConfig()` removed
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234587.
Bug: webrtc:7494
Change-Id: I251e03603394a4fc8769b9b5c197a157893676a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235060
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35206}
This change improves echo canceller transparency by enabling the use
of a non-capped ERLE when computing the residual echo spectrum for
dominant nearend detection.
Experimentation has shown that the feature improves echo canceller
transparency and user ratings.
Implementation CL:
https://webrtc-review.googlesource.com/c/src/+/221920
Bug: webrtc:12870
Change-Id: I7dc66810e8300cd35321bcd5b9fae9bc3386836d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234841
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35186}
Internal refactoring of AGC2. This CL is needed in preparation for its
child CL to correctly show the upcoming changes in the diff.
Bug: webrtc:7494
Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35170}
Move the check for analog gain changes so that it can be used
independently of echo_controller. This change is needed to land
https://webrtc-review.googlesource.com/c/src/+/234140.
Bug: webrtc:12774
Change-Id: I9ea127b0a4d374f31493d6f8afcacee40fa9257c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234383
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35159}
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).
Tested: compiled Chrome with this patch and made an appr.tc test call
Bug: webrtc:7494
Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35140}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
Instead of using two different headroom parameters, namely
`kHeadroomDbfs` and `kSaturationProtectorExtraHeadroomDb`, only use
the former that now also accounts for the deleted one - i.e., it equals
the sum of the two headrooms. In this way, tuning AGC2 will be easier.
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).
The unit tests changes in agc2/saturation_protector_unittest.cc are
required since `extra_headroom_db` is removed and the changes in
agc2/adaptive_digital_gain_applier_unittest.cc are required because
`AdaptiveDigitalGainApplier` depends on `kHeadroomDbfs` which has been
updated as stated above.
Bug: webrtc:7494
Change-Id: I0a2a710bbede0caa53938090a004d185fdefaeb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232905
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35109}
This CL improves `GainController2::CheckGainAdaptiveDigital`, namely:
- correctly initialize AGC2 with the correct number of channels
- attenuate the input signal in order to avoid that the target gain is
set to zero (which was the case before)
- run AG2 adaptive digital for a longer period to allow time to trigger
the adaptive behavior (namely, from 2s to 10s)
- minor code style improvements
Bug: webrtc:7494
Change-Id: Ib41de088b341bb30460238b83e306a507b2bc5af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233101
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35099}
Add histograms WebRTC.Audio.Agc.ClippingPredictor.Precision and WebRTC.Audio.Agc.ClippingPredictor.Recall. The histograms are defined in https://chromium-review.googlesource.com/c/chromium/src/+/3150271.
Bug: webrtc:12774
Change-Id: I1561ec7a61377c262f636d7aa3a5d5fd60a8839d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231460
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35083}
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.
Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
To focus on the ability to predict clipping, the clipping predictor
evaluator doesn't increment the true positive count anymore when a
prediction is simultaneously observed with a detection.
Note that `WebRTC.Audio.Agc.ClippingPredictor.F1Score` is still used
to log the F1 score - i.e., the histogram hasn't been renamed.
Bug: webrtc:12774
Change-Id: Ia987e568a6df2a3ddba7fa1b5697d6feda22d20c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231233
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34942}
Evaluate the clipping predictor whenever injected but keep using the
predictions only when allowed.
Bug: webrtc:12774
Change-Id: I9e8930a528d1d514d52b821a28b6c8ad0c3aeb5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231137
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34937}
Move Precision, Recall and F1-score computation from `AgcManagerDirect`
to a separate function that can be tested.
Bug: webrtc:12774
Change-Id: Iba20f153a72b7f957bf938e0642055d421045c02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231228
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34933}
This CL also includes the following changes:
- `AudioProcessing::Config::GainController2::noise_estimator`
deprecated
- `EnergyToDbfs()` optimized by removing unnecessary `sqrt`
- Unit test minor fix, incorrect type was used
Bug: webrtc:7494
Change-Id: I88a6672d6f7cd03fcf6a3031883522d256880140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230940
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34893}
Denormal numbers (see [1]) may origin in APM when the input is zeroed
after a non-zero signal. In extreme cases, instructions involving
denormal operands may run as much as 100 times slower, which seems to
be the case (to some extent) of crbug.com/1227566.
This CL adds a class that disables denormals only via hardware on x86
and on ARM. The class is used in APM and it is an adaption of [2].
Tested: appr.tc call on Chromium (Win, Mac)
[1] https://en.wikipedia.org/wiki/Denormal_number
[2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/platform/audio/denormal_disabler.h
Fixed: chromium:1227566
Change-Id: I0ed2eab55dc597529f09f93c26c7a01de051fdbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227768
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34701}
Acts as a compile time annotation, with corresponding run-time check
only when DCHECKs are enabled, and built using absl or pthreads mutexes.
Bug: None
Change-Id: Ie044c1ea1e576df71d634301f7df9d75cdf10b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226328
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34555}
As part of go/coil update code search links to not point to the
"master" branch.
Bug: chromium:1226942
Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34531}
Add a histogram WebRTC.Audio.Agc.InputClippingRate and logging of
max clipping rate in AgcManagerDirect.
Bug: webrtc:12774
Change-Id: I4a72119b65ad032fc50672e2a8fb4a4d55e1ff24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225264
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34450}
Integrate ClippingPredictorEvaluator in AgcManagerDirect adding the
possibility to run the predictor without affecting the analog gain
adjustment process.
The evaluator is used to compute precision, recall and F1 score.
F1 score and the measured clipping prediction intervals are logged as
`WebRTC.Audio.Agc.ClippingPredictor.F1Score` and `.PredictionInterval`
histograms respectively.
Bug: webrtc:12774
Change-Id: I708dcda9321f92d5bd17ec4c36ebce1165ead57f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221921
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34327}
The dominant nearend detector uses the residual echo spectrum for
determining whether in nearend state. The residual echo spectrum in
computed using the ERLE. To reduce the risk of echo leaks in the
suppressor, the ERLE is capped. While minimizing echo leaks, the
capping of the ERLE can affect the dominant nearend classification
negatively as the residual echo spectrum is often over estimated.
This change enables the dominant nearend detector to use a residual
echo spectrum computed with a virtually non-capped ERLE. This ERLE
is only used for dominant nearend detection and leads to increased
transparency.
The feature is currently disabled by default and can be enabled
with the field trial "WebRTC-Aec3UseUnboundedEchoSpectrum".
Bug: webrtc:12870
Change-Id: Icb675c6f5d42ab9286e623b5fb38424d5c9cbee4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221920
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34270}
This change enables the use of two different adaptation speeds of the
matched filter of the delay estimator of AEC3.
One speed is used when no delay has been found, and one is used after a
reliable delay has been found. The purpose is to use a slower adaptation
speed to reduce the risk of divergence during double-talk without
slowing down the search for the initial delay.
The CL prepares for experimentation by adding field trials for
controlling the two adaptation speeds.
Bug: webrtc:12775
Change-Id: I817a1ab5ded0f78d20de45edcf04c708290173fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219083
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34055}
This change adds the field trial "WebRTC-TransientSuppressorForcedOff"
that can be used to disable the transient suppressor (removal of
keyboard typing sounds). The field trial can be enabled by users via
command-line or via experimentation.
Bug: chromium:1186705
Change-Id: I7272df6a20fbbee24a7ba0904502c76bd775d275
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219282
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34038}
Update `AudioProcessing::Config::ToString()` to also dump the config
from `AnalogGainController` which is missing.
Bug: webrtc:7494
Change-Id: Iea5dab1f6abb9ec8581ce690a2a119f202b4d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219082
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34025}
This reverts commit 793bac569f.
Reason for revert: rare compilation error fixed
Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> > a) Handles for detached threads are simply closed where applicable.
> > b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
This reverts commit c89fdd716c.
Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
> a) Handles for detached threads are simply closed where applicable.
> b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=handellm@webrtc.org
Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
Add the option to run the adaptive digital controller of AGC2 without
side-effects - i.e., no gain applied.
Tested: adapation verified during a video call in chromium
Bug: webrtc:7494
Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33875}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
The change introduces support for detachable PlatformThreads, for which
the Stop() call doesn't wait until the thread has finished executing.
The change also introduces rtc::ThreadAttributes that carries priority
and detachability thread attributes. It additionally refactors all
known use to use the new semantics.
Bug: b:181572711, webrtc:12659
Change-Id: Id96e87c2a0dafabc8047767d241fd5da4505d14c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214704
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33796}
The ERLE is used to estimate residual echo for echo suppression. The
ERLE is reduced during far-end offset to avoid echo leakage. When there
is a strong near-end present this can cause unnecessary transparency loss.
This change adds an ERLE estimation that does not compensate for onsets and
uses it for residual echo estimation when the suppressor considers the near-end to be dominant.
Bug: webrtc:12686
Change-Id: Ida78eeacf1f95c6e62403f86ba3f2ff055898a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215323
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33786}
- Bug fix: the desired initial gain quickly dropped to 0 dB hence
starting a call with a too low level
- New tuning to make AGC2 more robust to VAD mistakes
- Smarter max gain increase speed: to deal with an increased threshold
of adjacent speech frames, the gain applier temporarily allows a
faster gain increase to deal with a longer time spent waiting for
enough speech frames in a row to be observed
- Saturation protector isolated from `AdaptiveModeLevelEstimator` to
simplify the unit tests for the latter (non bit-exact change)
- AGC2 adaptive digital config: unnecessary params deprecated
- Code readability improvements
- Data dumps clean-up and better naming
Bug: webrtc:7494
Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33736}
The current noise level estimator has a bug due to which the estimated
level decays to the lower bound in a few seconds when speech is observed.
Instead of fixing the current implementation, which is based on a
stationarity classifier, an alternative, lightweight, noise floor
estimator has been added and tuned for AGC2.
Tested on several AEC dumps including HW mute, music and fast talking.
Bug: webrtc:7494
Change-Id: Iae4cff9fc955a716878f830957e893cd5bc59446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214133
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33733}
Erle Uncertainty changes the residual echo computation during saturated
echo. However, the case of saturated echo is already handled by the
residual echo estimator causing the ErleUncertainty to be a no-op.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I779ba67f99f29d4475a0465d05da03d42d50e075
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215072
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33719}
Done in preparation for the child CL which adds an alternative
implementation.
Bug: webrtc:7494
Change-Id: I4963376afc917eae434a0d0ccee18f21880eefe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214125
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33646}
Code style improvements done in preparation for a bug fix (TODO added)
which requires changes in the unit tests.
Note that one expected value in the unit tests has been adjusted since
the white noise generator is now instanced in each separate test and
therefore, even if the seed remained the same, the generated sequences
differ.
Bug: webrtc:7494
Change-Id: I497513b84f50b5c66cf6241a09946ce853eb1cd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214122
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33636}
This is a reland of aa6adffba3
What was changed in the reland is that the merging of the bands is
excluded from the code that is not run when the output is not used.
I.e., the merging is always done.
This is important to have since some clients may apply muting before APM,
and still flag to APM that the signal is muted. If the merging is not
always done, those clients will get nonzero output from APM during muting.
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: Ib74dd1cefa173d45101e26c4f2b931860abc6d08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33478}
This CL adds functionality in the noise suppressor that allows the
computational complexity to be reduced when the output of APM is not used.
Bug: b/177830919
Change-Id: I849351ba9559fae770e4667d78e38abde5230eed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211342
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33477}
This CL adds functionality in AEC3 that allows the computational
complexity to be reduced when the output of APM is not used.
Bug: b/177830919
Change-Id: I08121364bf966f34311f54ffa5affbfd8b4db1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211341
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33476}
This CL adds functionality that allows adjusting the audio levels
internally in APM. The main purpose of the functionality is to allow
APM to optionally be moved to an integration that does not provide an
analog gain to control, and the implementation of this has been
tailored specifically to meet the requirements for that.
More specifically, this CL does
-Add a new variant of the pre-amplifier gain that is intended to replace
the pre-amplifier gain (but at the moment can coexist with that). The
main differences with the pre-amplifier gain is that an attenuating
gain is allowed, the gain is applied jointly with any emulated analog
gain, and that its packaging fits better with the post gain.
-Add an emulation of an analog microphone gain. The emulation is
designed to match the analog mic gain functionality in Chrome OS (which
is digital) but should be usable also on other platforms.
-Add a post-gain which is applied after all processing has been applied.
The purpose of this gain is for it to work well with the integration
in ChromeOS, and be used to compensate for the offset that there is
applied on some USB audio devices.
Bug: b/177830918
Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33466}
This CL adds one frame (10 ms) of silence in APM output after unmuting to mask
audio resulting from the turning on the processing that was deactivated
during the muting.
Bug: b/177830919
Change-Id: If44cfb0ef270dde839dcd3f0b98d1c91e81668dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33454}
This reverts commit aa6adffba3.
Reason for revert: breaks webrtc-importer
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: I937cd61dedcd43150933eb1b9d65aebe68401e91
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211348
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33433}
This CL selectively turns off parts of the audio processing when
the output of APM is not used. The parts turned off are such that
don't need to continuously need to be trained, but rather can be
temporarily deactivated.
The purpose of this CL is to allow CPU to be reduced when the
client is muted.
The CL will be follow by additional CLs, adding similar functionality
in the echo canceller and the noiser suppressor
Bug: b/177830919
Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33431}
Currently the echo canceller reference signal is high-pass filtered to
avoid the need of modeling the capture-side high-pass filter as part of
the echo path.
This can lead to the lowest frequency bins of the linear filter
diverging as there is little low-frequency content available for
training. Over time the filter can output an increasing amount of
low-frequency power, which in turn affects the filter's ability to
adapt properly.
Disabling the high-pass filtering of the echo canceller reference solves
this issue, resulting in improved filter convergence.
Bug: webrtc:12265
Change-Id: Ic526a4b1b73e1808cfcd96a8cdee801b96a27671
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208288
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33322}
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.
Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
This CL adds the initial support for letting APM know when its output
will be used or not.
It also adds a new method for passing RuntimeInformation to APM that
returns a bool indicating the success of the passing of information.
Bug: b/177830919
Change-Id: Ic2e1b92c37241d74ca6394b785b91736ca7532aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206061
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33201}
This CL adds the ability in audioproc_f and unpack_aecdump to:
-Clearly identify the Init events and when those occur.
-Optionally only process a specific Init section of an aecdump.
-Optionally selectively turn on dumping of internal data for a
specific init section, and a specific time interval.
-Optionally let unpack_aecdump produce file names based on inits.
Bug: webrtc:5298
Change-Id: Id654b7175407a23ef634fca832994d87d1073239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196160
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33181}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
This CL allows separate dump sets to be used when dumping internal
APM data using audioproc_f, opening up for reducing the amount of
data to be dumped.
Bug: webrtc:5298
Change-Id: I8286933ceed10db074f2064414cc08e8b12653fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196089
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33047}
- The injection of the AGC2 level estimator into `AgcManagerDirect`
is not used anymore
- `ExperimentalAgc::enabled_agc2_level_estimator` can also be removed
- 3 ctors of `ExperimentalAgc` are unused
- `AgcManagerDirectStandaloneTest::AgcMinMicLevelExperiment` can be
split into separate unit tests (better code clarity)
Bug: webrtc:7494
Change-Id: I5843147c38cf7cb5ee484b0a72fe13dcf363efaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202025
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33027}
This is a reland of 69241a93fb
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
residual_echo_detector and level_estimation is not part of gain_controller2
in AudioProcessing::Config.
Bug: None
Change-Id: Ic31b8c6bb4d9ed59b4f337e5dc8b8c587ae54202
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200924
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32972}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
Results: RNN VAD realtime factor improved from 140x to 195x (+55x)
Test device: Pixel 2 XL
Benchmark setup: max clock speed forced on all the cores by
setting "performance" as scaling governor
Bug: webrtc:10480
Change-Id: I3e92f643853ad1fe990db909c578ce78ee826c03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198842
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32888}
This change just affects the ERLE metric that is reported. The rest is unaffected and bitexact.
Bug: webrtc:12280
Change-Id: I2d28ef14a9b704c83aba18b624f67671eec4a042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196649
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32809}
The implementations for the fully connected layer can be simlpified by
using `VectorMath:DotProduct()`. In this way, it is also possible to
remove (nearly) duplicated SIMD code, reduce the binary size and more
easily maintain the code.
This CL also forces unoptimized code for the output layer of the VAD,
which is a FC 24x1 layer. A slight improvement of the realtime has
been measured (delta ~ +5x).
Bug: webrtc:10480
Change-Id: Iee93bd59f7905ebf96275dbbfeb3c921baf4e8db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195580
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32806}
Using `VectorMath::DotProduct()` in GatedRecurrentLayer to reuse existing
SIMD optimizations. Results:
- When SSE2/AVX2 is avilable, the GRU layer takes 40% of the unoptimized
code
- The realtime factor for the VAD improved as follows
- SSE2: from 570x to 630x
- AVX2: from 610x to 680x
This CL also improved the GRU layer benchmark by (i) benchmarking a GRU
layer havibng the same size of that used in the VAD and (ii) by prefetching
a long input sequence.
Bug: webrtc:10480
Change-Id: I9716b15661e4c6b81592b4cf7c172d90e41b5223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195545
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32803}
After the refined filter has been determined to perform better than
the coarse filter, and the coefficients of the coarse filters are
overwritten by the ones from the refined filter, at least 100 ms have
to pass before the adaptation of the refined filter is allowed to speed
up due to good coarse filter performance.
This change solves the vicious circle described in webrtc:12265, where
the coarse and refined filters can diverge over time.
This feature can be disabled remotely via a kill-switch. When disabled
the AEC output is bit-exact to before the change.
Bug: webrtc:12265,chromium:1155477
Change-Id: Iacd6e325e987dd8a475bb3e8163fee714c65b20a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196501
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32801}
This CL makes the static variable kChunkSizeMs constexpr to avoid
potential issues on some compilers.
Bug: None
Change-Id: I8e2b1819561ec96fb17d3899af95405cc36a6097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32795}
Handle the case in which neither WEBRTC_ARCH_X86_FAMILY nor
WEBRTC_HAS_NEON are defined.
Bug: webrtc:10480
Change-Id: I241583911d8e5645dfbd39b60337dd20b2d9f046
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196525
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32786}
This will soon become a compile-time error. Fix class hierarchies that
wrap StrictMock in a NiceMock or vice-versa by removing redundant
wrappings and removing inheritance from Nice/StrictMock and fixing the
call sites as appropriate.
Bug: b/173702213
Change-Id: Ic90b1f270c180f7308f40e52e358a8f6a6baad86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32783}
Refactoring done to more easily and cleanly add SIMD optimizations and
to remove `GatedRecurrentLayer` from the RNN VAD api.
Bug: webrtc:10480
Change-Id: Ie1dffdd9b19c57c03a0b634f6818c0780456a66c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195445
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32770}
Refactoring done to more easily and cleanly add SIMD optimizations and
to remove `FullyConnectedLayer` from the RNN VAD api.
Minor improvements (readability, API):
- `FullyConnectedLayer` gets the ActivationFunction enum and not
a function view anymore
- SSE2 optimization moved into `FullyConnectedLayer::ComputeOutputSse2`
- layer name added for improved logs
Bug: webrtc:10480
Change-Id: Ida4903a67655e19ef0464f378c433c1f6e96dca7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195444
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32766}
Plus a few minor code readability improvements.
Bug: webrtc:10480
Change-Id: I590d8e203b1d05959a8c15373841e37abe83237e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195334
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32764}
The HMM based transparent mode classifier is disabled until an issue
with diverging filters is resolved.
Bug: chromium:1155071
Change-Id: Iee249869f6ece1e48e834b3a4b9249c69a51286c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196341
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32756}
This CL brings a large improvement to the RNN VAD CPU performance
by finally using `VectorMath::DotProduct()` for pitch search.
The realtime factor improved from about 390x to 570x for SSE2
(+180x, 45% faster) and to 610x for AVX2 (+235x, 60% faster).
RNN VAD benchmark results:
```
+-----+-------+------+------+
| run | none* | SSE2 | AVX2 |
+-----+-------+------+------+
| 1 | 393x | 572x | 618x |
| 2 | 388x | 568x | 607x |
| 3 | 393x | 564x | 599x |
+-----+-------+------+------+
```
*: baseline, no SIMD used for pitch search, but SSE2 used for the RNN
Results obtained as follows:
1. Force SSE2 in `DISABLED_RnnVadPerformance` for the RNN part in
order to measure the baseline correctly:
```
RnnBasedVad rnn_vad({/*sse2=*/true, /*avx2=*/true, /*neon=*/false});
```
2. Run the test:
```
$ ./out/release/modules_unittests \
--gtest_filter=*RnnVadTest*DISABLED_RnnVadPerformance* \
--gtest_also_run_disabled_tests --logs
```
Bug: webrtc:10480
Change-Id: I89a2bd420265540026944b9c0f1fdd4bfda7f475
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195001
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32755}
This CL adds a new library for the RNN VAD that provides (optimized)
vector math ops. The scheme is the same of the `VectorMath` class of AEC3
to ensure correct builds across different platforms.
Bug: webrtc:10480
Change-Id: I96bcfbf930ca27388ab5f2d52c022ddb73acf8e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194326
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32741}
In preparation for adding AVX2 code, a safe scheme to support
different SIMD optimizations is added.
Safety features:
- AVX2 kill switch to stop using it even if supported by the
architecture
- struct indicating the available CPU features propagated from
AGC2 to each component; in this way
- better control over the unit tests
- no need to propagate individual kill switches but just
set to false features that are turned off
Note that (i) this CL does not change the performance of the RNN VAD
and (ii) no AVX2 optimization is added yet.
Bug: webrtc:10480
Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32739}
The output of GetLinearAecOutput is changed to have the range [-1, 1]
instead of [-2^15, 2^15] to be more similar to other Audio Processing
Module API functions.
The "--linear_aec_output" of audioproc_f has been tested for
bit-exactness.
Bug: webrtc:12185
Change-Id: Id50d93fcfaee5c239f3eb73f99d0bd3533319518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32604}
Add inverted lags index to simplify the loop in
`FindBestPitchPeriod48kHz()`. Instead of looping over 294 items,
only loop over the relevant ones (up to 10) by keeping track of
the relevant indexes.
The benchmark has shown a slight improvement (about +6x).
Benchmarked as follows:
```
out/release/modules_unittests \
--gtest_filter=*RnnVadTest.DISABLED_RnnVadPerformance* \
--gtest_also_run_disabled_tests --logs
```
Results:
| baseline | this CL
------+----------------------+------------------------
run 1 | 22.8319 +/- 1.46554 | 22.1951 +/- 0.747611
| 389.367x | 400.539x
------+----------------------+------------------------
run 2 | 22.4286 +/- 0.726449 | 22.2718 +/- 0.963738
| 396.369x | 399.16x
------+----------------------+------------------------
run 2 | 22.5688 +/- 0.831341 | 22.4166 +/- 0.953362
| 393.906x | 396.581x
This CL also moved `PitchPseudoInterpolationInvLagAutoCorr()`
into `FindBestPitchPeriod48kHz()`.
Bug: webrtc:10480
Change-Id: Id4e6d755045c3198a80fa94a0a7463577d909b7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191764
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32590}
This reverts commit 9da3e177fd.
Reason for revert: bug in ComputePitchPeriod48kHz()
Original change's description:
> RNN VAD: pitch search optimizations (part 1)
>
> TL;DR this CL improves efficiency and includes several code
> readability improvements mainly triggered by the comments to
> patch set #10.
>
> Highlights:
> - Split `FindBestPitchPeriods()` into 12 and 24 kHz versions
> to hard-code the input size and simplify the 24 kHz version
> - Loop in `ComputePitchPeriod48kHz()` (new name for
> `RefinePitchPeriod48kHz()`) removed since the lags for which
> we need to compute the auto correlation are a few
> - `ComputePitchGainThreshold()` was only used in unit tests; it's been
> moved into the anon ns and the test removed
>
> This CL makes `ComputePitchPeriod48kHz()` is about 10% faster (measured
> with https://webrtc-review.googlesource.com/c/src/+/191320/4/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc).
> The realtime factor has improved by about +14%.
>
> Benchmarked as follows:
> ```
> out/release/modules_unittests \
> --gtest_filter=*RnnVadTest.DISABLED_RnnVadPerformance* \
> --gtest_also_run_disabled_tests --logs
> ```
>
> Results:
>
> | baseline | this CL
> ------+----------------------+------------------------
> run 1 | 24.0231 +/- 0.591016 | 23.568 +/- 0.990788
> | 370.06x | 377.207x
> ------+----------------------+------------------------
> run 2 | 24.0485 +/- 0.957498 | 23.3714 +/- 0.857523
> | 369.67x | 380.379x
> ------+----------------------+------------------------
> run 2 | 25.4091 +/- 2.6123 | 23.709 +/- 1.04477
> | 349.875x | 374.963x
>
> Bug: webrtc:10480
> Change-Id: I9a3e9164b2442114b928de506c92a547c273882f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191320
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32568}
TBR=alessiob@webrtc.org,peah@webrtc.org
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10480
Change-Id: I2a91f4f29566f872a7dfa220b31c6c625ed075db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192660
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32581}
This reverts commit 2f7d1c62e2.
Reason for revert: bug in ancestor CL https://webrtc-review.googlesource.com/c/src/+/191320
Original change's description:
> RNN VAD: pitch search optimizations (part 2)
>
> This CL brings a large improvement to the VAD by precomputing the
> energy for the sliding frame `y` in the pitch buffer instead of
> computing them twice in two different places. The realtime factor
> has improved by about +16x.
>
> There is room for additional improvement (TODOs added), but that will
> be done in a follow up CL since the change won't be bit-exact and
> careful testing is needed.
>
> Benchmarked as follows:
> ```
> out/release/modules_unittests \
> --gtest_filter=*RnnVadTest.DISABLED_RnnVadPerformance* \
> --gtest_also_run_disabled_tests --logs
> ```
>
> Results:
>
> | baseline | this CL
> ------+----------------------+------------------------
> run 1 | 23.568 +/- 0.990788 | 22.8319 +/- 1.46554
> | 377.207x | 389.367x
> ------+----------------------+------------------------
> run 2 | 23.3714 +/- 0.857523 | 22.4286 +/- 0.726449
> | 380.379x | 396.369x
> ------+----------------------+------------------------
> run 2 | 23.709 +/- 1.04477 | 22.5688 +/- 0.831341
> | 374.963x | 393.906x
>
> Bug: webrtc:10480
> Change-Id: I599a4dda2bde16dc6c2f42cf89e96afbd4630311
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191484
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32571}
TBR=alessiob@webrtc.org,peah@webrtc.org
Change-Id: I53e478d8d58912c7a5fae4ad8a8d1342a9a48091
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10480
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192620
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32580}
This reverts commit ea89f2a447.
Reason for revert: bug in ancestor CL https://webrtc-review.googlesource.com/c/src/+/191320
Original change's description:
> RNN VAD: pitch search optimizations (part 3)
>
> `ComputeSlidingFrameSquareEnergies()` which computes the energy of a
> sliding 20 ms frame in the pitch buffer has been switched from backward
> to forward.
>
> The benchmark has shown a slight improvement (about +6x).
>
> This change is not bit exact but all the tolerance tests still pass
> except for one single case in `RnnVadTest,PitchSearchWithinTolerance`
> for which the tolerance has been slightly increased. Note that the pitch
> estimation is still bit-exact.
>
> Benchmarked as follows:
> ```
> out/release/modules_unittests \
> --gtest_filter=*RnnVadTest.DISABLED_RnnVadPerformance* \
> --gtest_also_run_disabled_tests --logs
> ```
>
> Results:
>
> | baseline | this CL
> ------+----------------------+------------------------
> run 1 | 22.8319 +/- 1.46554 | 22.087 +/- 0.552932
> | 389.367x | 402.499x
> ------+----------------------+------------------------
> run 2 | 22.4286 +/- 0.726449 | 22.216 +/- 0.916222
> | 396.369x | 400.162x
> ------+----------------------+------------------------
> run 2 | 22.5688 +/- 0.831341 | 22.4902 +/- 1.04881
> | 393.906x | 395.283x
>
> Bug: webrtc:10480
> Change-Id: I1fd54077a32e25e46196c8e18f003cd0ffd503e1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191703
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32572}
TBR=alessiob@webrtc.org,kwiberg@webrtc.org
Change-Id: I57a8f937ade0a35e1ccf0e229c391cc3a10e7c48
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10480
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192621
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32578}
