Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: I7cde835161e2d3e85fc7c919556fa9a9e87ef6df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238169
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35393}
New limiter tuning to more quickly go back to 0 dB after the limiter
kicks in and the input peak level goes back to normal.
Bug: webrtc:7494
Change-Id: I1050957ca4caf12c4562b899b16c306957dce169
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237701
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35384}
Reducing pointer following. This will allow the compiler to optimize more efficiently with the "-fno-strict-aliasing" flag.
Bug: None
Change-Id: Ib1fd3a1cf3f89471b0ec87404650a6061eec5e2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Christian Schuldt <cschuldt@google.com>
Cr-Commit-Position: refs/heads/main@{#35374}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
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}
