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}
`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}
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}
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}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
Avoid excessive echo suppression in frequencies above 2 kHz when
there is a dominant nearend. Calls with clock drift will not be affected
by this change as they tend to have less accurate linear filters.
Bug: webrtc:11985
Change-Id: Iddc628da5e2ba572c1b47acd87dd3be35260dca1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188580
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32533}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
This CL includes two changes:
1. the type for (inverted) lags and pitch periods changed from
size_t to int to reduce the chance of bugs with pitch period
manipulations
2. CheckLowerPitchPeriodsAndComputePitchGain() is optimized by
replacing an unnecessary if statement inside the loop with the
predetermined number of loops
Bug: webrtc:10480
Change-Id: I38432699254b37a2c0111279c28be8dc65b87e9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139252
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32521}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This CL removes the circular shift to compute the convolution in
`ComputeLpResidual()`, which is now 4x faster (benchmarked with the
`RnnVadTest.DISABLED_ComputeLpResidualBenchmark` unit test).
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Ia7767d9b57378c12c8ff31f58fea03905be5c5de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32491}
Use hard-coded constants previously computed at runtime within a loop in
`DenoiseAutoCorrelation`.
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: I02c2fff7dc7153ea2ab8a27cad8a479a0f029996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189963
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32489}
The cross-correlation function in the anonimous namespace of
lp_residual.cc is used once as to compute auto-correlation coefficients.
Hence, it can be simplified (and renamed).
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Ie769e42907c15572172a8ad1a34e82ce60c4a386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189962
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32464}
Use hard-coded constants previously computed at runtime within a loop in
`ComputeAndPostProcessLpcCoefficients`.
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Iaec178c56937449c3fe9af881354d19cc8c47c7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189961
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32462}
This change makes the transparent mode classifier consider filter
convergence also for microphone signals with very low volume. This is
in order to prevent entering transparent mode when there is low, but
audible, echo.
Furthermore, the suppression gain during transparent mode is restored
to avoid leaks when the echo is too low to be reliably detected by
filter convergence.
Bug: webrtc:10232, chromium:1140452
Change-Id: Idd4f40c4aee7c20baa444afaa5ec604eb65bcfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189786
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32461}
These metrics are no longer in use:
WebRTC.Audio.EchoCanceller.ActiveRender
WebRTC.Audio.EchoCanceller.ErleBand*
WebRTC.Audio.EchoCanceller.ErlBand*
Corresponding CL to mark obsolete in histograms.xml:
https://chromium-review.googlesource.com/c/chromium/src/+/2485059
Bug: chromium:1140067,chromium:1140071,chromium:1140073
Change-Id: Icb79b540f340a596c178f352363b30dc7b885251
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189783
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32449}
AGC2 is correctly (re)initialized when its config changes.
This CL also improves the `AudioProcessingImpl::ApplyConfig`
readability by defining operator!= also for the AGC1 config.
Bug: webrtc:7494
Change-Id: I62068de32c941e6b18d4618c656f569647042345
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32402}
The APM config to string mapping must be in one place (namely,
in `audio_processing.cc`). This CL moves the AGC2 config to string
impl to the right place.
This CL also updates `GainController2::Validate()` and adds the
missing unit tests for the parameters that have recently been added.
Stack buffer size in `AudioProcessing::Config::ToString()` increased
because of the extra params. Syntax near `multi_channel_capture` fixed.
Output string format verified with a JS linter.
Bug: webrtc:7494
Change-Id: I692e1549b7d40c970d88a14c8e83da16325fb54c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187080
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32400}
This CL adds delay headroom when an external delay estimator is used.
Tested: audioproc_f --aec=1 is bitexact on a large number of aecdumps
Bug: b/158455753
Change-Id: I56de44e841bb8162e302181c6c386ad7fbb00dee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186703
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32323}
This is a reland of 2978abb88c
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
Bug: b/150764764,chromium:1134939
Change-Id: I5eea60b35e6d09003ec2fee3865513df8bdd5823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32304}
This is a reland of ad148272b8
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663, chromium:1134234
Change-Id: I0cb34cf08d4d14bc3aee055254493c9c9ee8faa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186401
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32303}
This adds a config parameter for reverb modelling in nonlinear mode.
The killswitch is preserved, and will have the same effect before and after this CL.
Tested: Bitexact on a large number of aecdumps both with and without WebRTC-Aec3NonlinearModeReverbKillSwitch.
Bug: webrtc:8671
Change-Id: I7b2fa92307af5316dbe895392367abb8d15f06bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183541
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32298}
New params have recently been added, but ToString() was not updated.
Bug: webrtc:7494
Change-Id: I2115ccb08d29e860a658284d096e48da17daaedd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186267
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32283}
This reverts commit 18515e2793.
Reason for revert: Continuation of previous revert https://webrtc-review.googlesource.com/c/src/+/186304
Original change's description:
> Reland "Reduce the amount of howling reduction in AEC3"
>
> This is a reland of 2978abb88c
>
> Original change's description:
> > Reduce the amount of howling reduction in AEC3
> >
> > This CL backs off the howling protection functionality in AEC3.
> > The effect is increased transparency in some cases. No negative effects
> > have been identified in the hands-on testing.
> >
> >
> > A kill-switch is added that can be used to turn off the functionality.
> >
> > Bug: b/150764764
> > Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32258}
>
> Bug: b/150764764
> Change-Id: I277f24eb3288ad0307e7463bad9aea6436cfe879
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186263
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32274}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I0596855e3bf910878aebcee90e03345d5db1105d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/150764764
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186269
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32276}
This is a reland of 2978abb88c
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
Bug: b/150764764
Change-Id: I277f24eb3288ad0307e7463bad9aea6436cfe879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186263
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32274}
This is a reland of ad148272b8
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663
Change-Id: Ib41dc1d1c5865f2828699c462939d15d5562df47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186262
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32270}
This CL adds and wires up the following parameters:
- VAD probability attack used in `VadLevelAnalyzer`
- Adjacent spech frames threshold used in `AdaptiveModeLevelEstimator`
- Initial saturation margin used in `AdaptiveModeLevelEstimator`
The deprecated ctor in `AdaptiveModeLevelEstimator` is removed.
Tested: bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Idf94aaadba1476757f845e696bfb47ff6252d5f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186048
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32265}
This CL adds and wires up a parameter (namely, adjacent speech
frames threshold) used in `AdaptiveDigitalGainApplier`.
Tested: bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: I751cd91f08a6e98ee20f767c8df0ed121c8d4b68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186049
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32264}
Make the digital adaptive gain applier more robust to VAD false
positives. Achieved by allowing a gain increase only if enough adjacent
speech frames are observed.
Tested:
- Bit-exactness verified with audioproc_f
- If `kDefaultDigitalGainApplierAdjacentSpeechFramesThreshold` == 2
then not bit-exact
Bug: webrtc:7494
Change-Id: I3bab5a449aaf0ef1a64b671b413ba2ddb4688cd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32263}
This reverts commit 2978abb88c.
Reason for revert: Breaks ApmTest.Process.
After trying to enable AVX2, we increased the amount of error
we tolerate (kFloatNear) and this CL introduced a regression which
makes the test fail after we reverted the enabling of AVX2 (restoring
the old tolerance).
With this CL:
../../modules/audio_processing/audio_processing_unittest.cc:1779: Failure
The difference between test->rms_dbfs_average() and rms_dbfs_average is 0.00142669677734375, which exceeds kFloatNear, where
test->rms_dbfs_average() evaluates to 52.907142639160156,
rms_dbfs_average evaluates to 52.905715942382812, and
kFloatNear evaluates to 0.00050000000000000001.
[ FAILED ] ApmTest.Process (5347 ms)
[----------] 1 test from ApmTest (5348 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test suite ran. (5350 ms total)
[ PASSED ] 0 tests.
[ FAILED ] 1 test, listed below:
[ FAILED ] ApmTest.Process
After reverting it:
[ OK ] ApmTest.Process (5345 ms)
[----------] 1 test from ApmTest (5347 ms total)
[----------] Global test environment tear-down
[==========] 1 test from 1 test suite ran. (5350 ms total)
[ PASSED ] 1 test.
Original change's description:
> Reduce the amount of howling reduction in AEC3
>
> This CL backs off the howling protection functionality in AEC3.
> The effect is increased transparency in some cases. No negative effects
> have been identified in the hands-on testing.
>
>
> A kill-switch is added that can be used to turn off the functionality.
>
> Bug: b/150764764
> Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32258}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I4723c5cd66e3046851089157ec586afab55c5ce8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/150764764
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186122
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32261}
This reverts commit ad148272b8.
Reason for revert: Speculative revert to investigate test failures
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
TBR=mbonadei@webrtc.org,saza@webrtc.org,peah@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11663
Change-Id: Ibb019e8c702dce45ebf47f1c1e8db19069b4964d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32259}
This CL backs off the howling protection functionality in AEC3.
The effect is increased transparency in some cases. No negative effects
have been identified in the hands-on testing.
A kill-switch is added that can be used to turn off the functionality.
Bug: b/150764764
Change-Id: I604c569c76f911799556a60bc8fd2fb43bbfe196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186082
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32258}
This CL was written in preparation for the next CL in the chain and
it contains the following changes:
- SignalWithLevels -> AdaptiveDigitalGainApplier::FrameInfo
- Frame view removed from AdaptiveDigitalGainApplier::FrameInfo
- AdaptiveDigitalGainApplier::Process now gets side info as const& to
avoid unnecessary copies
- AdaptiveAgc::Process: `last_audio_level` renamed to `limiter_envelope`
to better reflect what that actually is
- Missing class/method docstrings added
Tested: bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Ie25dcd389d6eed74ea9a65f0720eeb8f20f0096b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186040
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32251}
This is the last CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.
This CL adds a second state property to hold temporary updates and a
counter for consecutive speech frames. When enough speech frames are
observed, the reliable state is updated; otherwise, the temporary state
is discarded.
The default for `AdaptiveModeLevelEstimator::min_consecutive_speech_frames_`
is 1, which means that the new feature is disabled.
Tested:
- Bit-exactness verified with audioproc_f
- Not bit-exact if `min_consecutive_speech_frames_` set to 10
Bug: webrtc:7494
No-Try: True
Change-Id: I0daa00e90c27c418c00baec39fb8eacd26eed858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185125
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32250}
- State -> LevelEstimatorState
- Mark two methods as const
- Call DumpDebugData() in one place
- DumpDebugData: don't check if data dumper is provided
- Add LevelEstimatorState::operator==
The changes will reduce clutter in follow up CL.
Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".
Bug: webrtc:7494
Change-Id: If39ce4b787069bef4af910d718cdfae3af1784a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185811
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32247}
`AdaptiveModeLevelEstimator::last_level_dbfs_` doesn't need to be optional.
Note: this CL breaks the chain of 3 CLs titled
"AGC2 AdaptiveModeLevelEstimator min consecutive speech frames".
Bug: webrtc:7494
Change-Id: Id5b409ca5cb5f11ed132c861b7995b9721e167bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185809
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32237}
This is the second CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.
In this CL, the `SaturationProtector` class has been replaced by a
struct that define the state and two functions to change it.
This is done in order to use the saturation protector state in
`AdaptiveModeLevelEstimator::State` and will allow to add a
temporary state in `AdaptiveModeLevelEstimator` (see the child CL).
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Ic5ecd1e174010656ed20664ef7b7e5798ebb7978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185041
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32226}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
This change completely disables the use of suppression
when operating in transparent mode.
It also removes the following field trials:
* WebRTC-Aec3UseLowEarlyReflectionsTransparentModeGain
* WebRTC-Aec3UseLowLateReflectionsTransparentModeGain
Bug: webrtc:11985
Change-Id: I1c75efdad2d9c9d0a1aced86bf0278fc96616ea1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185402
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32223}
This is a reland of ad148272b8
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
Bug: webrtc:11663
Change-Id: I669435c2f4e451ee0766d809443484f2dde09d8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32200}
This is the first CL needed to add a new `AdaptiveModeLevelEstimator`
feature that makes AGC2 more robus to VAD mistakes: the level estimator
discards estimation updates when too few consecutive speech frames are
observed.
In this CL, the state of the estimator is defined in a separate struct
so that in a follow-up CL a new member of that type can be added to
hold a temporary state (that can be either confirmed or discarded).
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: Ic2ea5ed63c493b9f3a79f19e7f5eaecaa6808ace
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184931
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32199}
Feature added to gain robustness to occasional VAD speech probability
spikes. In such a case, the attack process reduces the chance that the
smoothed values are greater than the speech threshold.
Bug: webrtc:7494
Change-Id: I6babe5afe30ea3dea021181a19d86bb74b33a98c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185046
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32198}
This reverts commit ad148272b8.
Reason for revert: Causing test failures downstream.
Original change's description:
> Activating AVX2 support by default
>
> This CL activates the newly added AVX2 support by default.
> The activation is done beneath a kill-switch.
>
> Beyond the above, the CL also changes an incorrect DCHECK_GT
> to a DCHECK_GE.
>
> Bug: webrtc:11663
> Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32193}
TBR=mbonadei@webrtc.org,saza@webrtc.org,peah@webrtc.org,kwiberg@webrtc.org
Change-Id: If2287a0a4b37931ce5f85baae093a66b19d0a78b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185481
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32196}
Refactoring CL to improve names and allow to inject a VAD into
`VadLevelAnalyzer` (new name for `VadWithLevel`).
The injectable VAD is needed to inject a mock VAD and write better
unit tests as new features are going to be added to the class.
Bug: webrtc:7494
Change-Id: Ic0cea1e86a19a82533bd40fa04c061be3c44f068
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185180
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32195}
This CL activates the newly added AVX2 support by default.
The activation is done beneath a kill-switch.
Beyond the above, the CL also changes an incorrect DCHECK_GT
to a DCHECK_GE.
Bug: webrtc:11663
Change-Id: I231ccb2f5efabf74cd8190411daa954b2b94a2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183042
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32193}
First step to only expose the relevant RNN VAD API to AGC2.
Bug: webrtc:7494
Change-Id: I7f11f6eebded124c30cabd64963c8e3ccc35e58f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185124
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32185}
This change introduces a new Hidden Markov Model based classifier for
AEC3's 'transparent mode'. Transparent mode is used with
headsets/headphones where the speaker signal does not leak into the
microphone signal.
The current classifier suffers from two problems:
1. It sometimes takes a long time to enter transparent mode.
2. Sometimes transparent mode is left (and it once again takes a long
time to re-enter).
Both problems have a severe effect on AEC transparency.
The new classifier enters transparent mode quicker and is less likely
to exit transparent mode when there is no echo. This improves the
audio experience when using headset/headphones.
Another (minor) benefit of this change is that when transparent mode
is disabled no classifier is run (or even created) saving some memory
and CPU cycles.
Bug: webrtc:10232
Change-Id: I509af0e22b59463aeaead53c78c35be1e97fe8c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32182}
In preparation for a coming refactoring CL, the (fixed) extra saturation
margin is now applied into `AdaptiveModeLevelEstimator`.
This CL also improves the unit tests by hard-coding its saturation
params instead of reading them from a field trial.
This reduces the chances of making the test flaky if a default value
changes.
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: I6765def9887a2f4e55b04d929af754cfecbb1626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184927
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32172}
- Passing the speech peak power instead of VAD data
- The private class SaturationProtector::PeakEnveloper has been removed
- Added `initial_saturation_margin_db_` parameter to correctly
initialize `last_margin_` (renamed to `margin_db_`)
- Member names have been fixed and/or shortened for better readability
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: I6cad2974397319737c8ac201d44311bf16275f28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184925
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32168}
Even if small, the peak delay buffer copies N-1 elements for each frame
whereas a ring buffer is copy-free and scales better if the buffer size
increases.
Tested: Bit-exactness verified with audioproc_f
Bug: webrtc:7494
Change-Id: If8c33877b7ab1d881a0606e222b26857a82fff69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184920
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32165}
This CL adds explicit initialization of the FilterAnalyzer in AEC3.
While the current code never uses any fields before they are initialized,
it makes sense to be on the safe side and add initialization during
construction.
Bug: webrtc:11918
Change-Id: I467c4c8b8d6dd859a1b216baef28ac1e9d3f76c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183764
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32069}
This is a reland of 2b242d8fba
Original change's description:
> Merge cpu_features build targets into //system_wrappers.
>
> Before this CL, functions declared in cpu_features_wrapper.h where
> not defined in the same build target, causing brittle builds that
> might fail at link time if the binary was not depending on
> //system_wrappers (the target with the definitions), violating [1].
>
> This CL moves everything into //system_wrappers and also moves
> cpu_features_wrapper.h definitions from C to C++ (in order to be able
> to add the definitions to a C++ build target like //system_wrappers).
>
> [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
>
> Bug: None
> Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32039}
TBR=kwiberg@webrtc.org
Bug: None
Change-Id: I1695b9a34d3ec20c50c1202a745f64fac58edef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183444
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32042}
This reverts commit 2b242d8fba.
Reason for revert: Breaks downstream project.
Original change's description:
> Merge cpu_features build targets into //system_wrappers.
>
> Before this CL, functions declared in cpu_features_wrapper.h where
> not defined in the same build target, causing brittle builds that
> might fail at link time if the binary was not depending on
> //system_wrappers (the target with the definitions), violating [1].
>
> This CL moves everything into //system_wrappers and also moves
> cpu_features_wrapper.h definitions from C to C++ (in order to be able
> to add the definitions to a C++ build target like //system_wrappers).
>
> [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
>
> Bug: None
> Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32039}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I4daa7582e55a0343eef72f08ed023c73e0b6456b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32040}
Before this CL, functions declared in cpu_features_wrapper.h where
not defined in the same build target, causing brittle builds that
might fail at link time if the binary was not depending on
//system_wrappers (the target with the definitions), violating [1].
This CL moves everything into //system_wrappers and also moves
cpu_features_wrapper.h definitions from C to C++ (in order to be able
to add the definitions to a C++ build target like //system_wrappers).
[1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs
Bug: None
Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32039}
This CL removes the possibility that APM cannot be created, i.e., that
the create method can return nullptr. That was already the case
implicitly but this CL makes that behavior explicit.
Bug: webrtc:5298
Change-Id: I2706ea538c9d1b4bcd65faecab637640a209a4dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183101
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32029}
This is a reland of 8be2f201ba
Original change's description:
> Add ability to state whether the APM output will be used
>
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
>
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}
Bug: b/154437967
Bug: b/163802450
Change-Id: Ia77a9e43f913929d1afa72212f1ea6c192d0e519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181887
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31957}
This reverts commit 8be2f201ba.
Reason for revert: Breaks downstream
Original change's description:
> Add ability to state whether the APM output will be used
>
> This CL adds the ability for the surrounding code to state that the
> APM output will not be used. The intended usecase for this is to allow
> APM to run at a lower complexity when the endpoint is muted.
> When APM has been informed that the output will not be used, it can
> turn off code that is needed only for ensuring that the output audio
> will sound good.
>
> Bug: b/154437967,b/163802450
> Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31949}
TBR=alessiob@webrtc.org,peah@webrtc.org
Change-Id: I1e56dafbbfa6ea69cccbbb5cdc2b1e2a6c122c11
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/154437967
Bug: b/163802450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181884
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31953}
This CL adds the ability for the surrounding code to state that the
APM output will not be used. The intended usecase for this is to allow
APM to run at a lower complexity when the endpoint is muted.
When APM has been informed that the output will not be used, it can
turn off code that is needed only for ensuring that the output audio
will sound good.
Bug: b/154437967,b/163802450
Change-Id: I8e22989e35354372e96191d15da44beb9d1b26ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181200
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31949}
Internal counters in the RenderDelayBuffer can slip out of sync with external counters, leading to buffer misalignment.
This CL gives the RenderDelayBuffer an opportunity to update its counters.
Tested:
Passes: modules_unittests --gtest_filter=BlockProcessor.*
Fails as expected due to new unit test: modules_unittests --gtest_filter=BlockProcessor.* --force_fieldtrials="WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch/Enabled/"
audioproc_f with default AEC settings has been verified to be bit-exact on a large number of aecdumps.
Bug: webrtc:11803
Change-Id: I9363b834c8c8c934add0335013df60bf131da4bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180126
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31795}
Removes usage of Chromium's //third_party/pymock in favor of the version
provided by vpython. This is so that the third_party version can
eventually be removed.
TBR=aleloi@webrtc.org
Bug: chromium:1094489
Change-Id: I68511e11ed1e517c2b6d3bb832090a3c27e480e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@google.com>
Cr-Commit-Position: refs/heads/master@{#31568}
This required for gcc-10.
Bug: None
Change-Id: Iba3ff0881005cb2ae4574e47284a1b881594de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176361
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31417}
This CL extends the WebRTC testing API to allow audioproc_f -based
testing using a pre-created AudioProcessing object. This is an
important feature to allow testing any AudioProcessing objects
that are injected into WebRTC.
Beyond adding this, the CL also changes the simulation code to
operate on a scoped_refptr<AudioProcessing> object instead of a
std::unique<AudioProcessing> object
Bug: webrtc:5298
Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31319}
...for the NullDataDumper, WrongCaptureBlockSize and
DISABLED_WrongRenderBlockSize tests. This is to avoid creation
of additional threads on Mac, which can cause issues on asan bots.
Bug: webrtc:11577
Change-Id: I4e6a64d47ec3b0a0e0018b19a0486208ba7e6ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175600
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31307}
This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.
This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.
The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.
The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
surface of APM.
2) Those files anyway needed to be moved to a separate build-
target to avoid a circular build-file dependency caused by
the other changes in this CL
Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}
This API is and has always been unused.
Bug: webrtc:5298
Change-Id: If1201d37a00e387567d44a9ed8be99a157915b47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174661
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31180}
This makes it optional for an echo detector to report metrics through
the getStats interface.
Bug: webrtc:11539
Change-Id: I1fef93b7bf534637b69c16971d38709b3e849a08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31146}
This introduces a function AudioProcessingImpl::SetCreateOptionalSubmodulesForTesting to simulate the exclusion of build-optional submodules, and tests of the currently only excludable submodule.
Bug: webrtc:11292
Change-Id: If492606205c9fdc669a6dce3a8989a434aeeed1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173746
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31138}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
This CL moves the two OOURA implementations present in the WebRTC tree
in one place.
No functional change is expected.
TBR=kwiberg@webrtc.org
No-Try: True
Bug: webrtc:11509
Change-Id: I330a9ec57e3dc65c9c8b43edd4bb295c55920efa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173682
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31106}
This is a reland of 12e2d4ddb2
Original change's description:
> APM: Remove the usage of AudioFrame in the AudioProcessing interface
>
> This CL removes the AudioFrame-based APIs from the AudioProcessing
> interface.
>
> Bug: webrtc:5298
> Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31016}
Bug: webrtc:5298
Change-Id: I70e6d59afc3716ee6109d8b9dc384abc71c93624
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173476
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31066}
This CL removes the AudioFrame-based APIs from the AudioProcessing
interface.
Bug: webrtc:5298
Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31016}
This CL adds several field-trial-based overrides for parameters related
to AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11487,chromium:1067597
Change-Id: Ica9613db74240687fc85efe059874ef8c20aa7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172844
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31007}
This CL adds two kill-switches to the AEC3 code to be used as
safe fallbacks to increase AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11487,chromium:1067597
Change-Id: I7f9f78db4964990bcdfa9adae6ef36a56bce7224
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172840
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30998}
This allows clients to exclude the transient suppression submodule from WebRTC builds, by defining WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR.
The changes have been shown to be bitexact for a test dataset (when the flag is _not_ defined.)
No-Try: True
Bug: webrtc:11226, webrtc:11292
Change-Id: I6931c82a280a9b40a53ee1c2a9820ed9e674a9a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171421
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30978}
This allows the user to run audioproc_f with various field trials set.
The approach is copied from test/test_main_lib.cc.
Tested:
1. Verified bitexactness vs ToT audioproc_f on a large dataset of aecdumps
2. Ran it with flags --aec=1 --force_fieldtrials="WebRTC-Aec3ClampInstQualityToZeroKillSwitch/Enabled/WebRTC-Aec3ClampInstQualityToOneKillSwitch/Enabled/" and verified in GDB that the AEC3 config was changed accordingly.
No-Try: True
Bug: webrtc:5298
Change-Id: I70eec7777f70893b36af33794a5842f67d56af31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172623
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30976}
Currently, audioproc_f crashes on a DCHECK as the data vector of Int16Frame is not resized.
Bug: webrtc:5298
Change-Id: I897cf0fce07e0ed2c0a365a965fa50fd3d8ddd18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172624
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30969}
This CL adds a number of kill-switches to the AEC3 code to be used as
safe fallbacks to increase AEC transparency.
The changes have been shown to be bitexact for a test dataset.
Bug: webrtc:11475,chromium:1066836
Change-Id: Ibebcbbfbbd958cb6fcc6993247e3030fa65b582c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172600
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30964}
This started to happen after turning on "gn analyze" on trybots. It
looks like this code was never built on MSVC trybots.
This CL tries to avoid the type deduction.
Error:
quality_assessment/sound_level.cc(103):
error C3535: cannot deduce type for 'const auto *' from '_FwdIt'
with
[
_FwdIt=std::_Array_iterator<int16_t,1440>
]
Bug: webrtc:11262
Change-Id: Iea7cf2ec62f1d0edfcf6ceac169c92050339d3c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172088
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30933}
This moves rms_level.* into a separate build target.
Bug: webrtc:11226
Change-Id: I94ceacd1ec65dda48f5d19b22ba2625d13543e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170323
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30856}
The error paths free the memory referenced by each pointer in the
struct, but if the pointers are not initialized, random memory belonging
to other parts of the program could be freed instead. Zero out the
entire struct as soon as it is allocated to ensure that nothing is freed
if there is nothing to free.
Bug: webrtc:11446
Change-Id: I8a2985d1388477339351aa03107ee68925372d49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171121
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30852}
This CL adds namespaces to those files remaining within APM that do not
have any such.
BUG=webrtc:5298
Change-Id: I710b3d2a3644bea9d4bdffef0d77883b30303338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171111
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30850}
This CL changes so that the AGC legacy C code is built as C++.
The CL also
-removes #defines from the header files.
-adds namespaces
-removes unused code.
To simplify the review, the CL is partitioned into different patchsets
where each comprising of one step in the modification of the code
(e.g., patch set 1 performs the renaming of the .c files to .cc).
Bug: webrtc:5298
Change-Id: I362b17bde91142b2f2166acba4f2f888efd50fa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171064
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30847}
This CL renames the shadow filter in AEC3 to have the more accurate name
coarse filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "shadow" with "coarse" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: I28d6041d0d34e85f8f8048d004b44a1a5f07bb07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170981
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30846}
This CL renames the main filter in AEC3 to have the more accurate name
refined filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "main" with "refined" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: Ifd0aab34e291736a2250e0986348404618630b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170825
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30843}
A follow-up CL will move the rms_level.* files into the new target.
Bug: webrtc:11226
Change-Id: I59579b026346f627c0a2739d25f90c12bffbf248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171102
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30840}
This CL removes the redundant VAD output from the newly introduced
integer API in AudioProcessing.
Bug: webrtc:5298
Change-Id: Iad2b1b97ada7f4863139655526c110e326c6788a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170824
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30832}
This CL replaces all remaining usage of AudioFrame within APM,
with the exception of the AudioProcessing interface.
The main changes are within the unittests.
Bug: webrtc:5298
Change-Id: I219cdd08f81a8679b28d9dd1359a56837945f3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170362
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30831}
This CL corrects an issue in the storing of the processed capture output
into aecdump recordings for the case when the integer API interface is
used.
Bug: webrtc:11441
Change-Id: I24aad47b5d62e0738d412ec270ad1db3a76aa94f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170823
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30829}
This CL moves the implementation of of the AudioFrame
support from the implementation of AudioProcessing
to proxy methods that map the call to the integer
stream interfaces (added in another CL).
The CL also changes the WebRTC code using the AudioFrame
interfaces to instead use the proxy methods.
This CL will be followed by one more CL that removes
the usage of the AudioFrame class from the rest of
APM (apart from the AudioProcessing API).
Bug: webrtc:5298
Change-Id: Iecb72e9fa896ebea3ac30e558489c1bac88f5891
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170110
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30812}
This CL creates a new stream interface and uses it to replace
most of the usage of AudioFrame in the non-test code.
The CL changes some of the test code as well, as the other
changes required that.
The CL will be followed by 2 more related CLs.
Bug: webrtc:5298
Change-Id: I5cfbe6079f30fc3fbf35b35fd077b6fb49c7def0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170040
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30799}
This is a first step to make the transient suppressor and voice detection optional.
Bug: webrtc:11226, webrtc:11292
Change-Id: I203125e11694a957a32bc7f98f3bec3ec8867839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166523
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30783}
This CL fixes the build for the meta taret "all"
(ninja -C out/Debug all).
More interestingly fixes cascaded_biquad_filter_unittest.cc which
seems not to be run at the moment.
Bug: webrtc:11411
Change-Id: I3d5f83c3898cca96aff8fbdad97d7b48caa9fffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169858
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30722}
This CL refactors and optimizes the 3-band split-filter in APM, which
is a very computationally complex component.
Beyond optimizing the code, the filter coefficients are also quantized
to avoid denormals.
The changes reduces the complexity of the split filter by about 30-50%.
The CL has been tested for bitexactness on a number of aecdump
recordings.
(the CL also removes the now unused code for the sparse_fir_filter)
Bug: webrtc:6181
Change-Id: If45f8d1f189c6812ccb03721156c77eb68181211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168189
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30592}
This CL avoids the head-allocations done in a sum of the squared values
in a nested vector.
Bug: webrtc:11361, chromium:1052086
Change-Id: I698b855bdd54df2147ef3b6d5e3d401401228d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168543
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30520}
This CL removes the code for the deprecated legacy noise.
Bug: webrtc:5298
Change-Id: If287d8967a3079ef96bff4790afa31f37d178823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167922
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30434}
This CL adds support for reading and writing floating point
wav files in WebRTC. It also updates the former wav handling
code as well as adds some simplifications.
Beyond this, the CL also adds support in the APM data_dumper
and in the audioproc_f tool for using the floating point wav
format.
Bug: webrtc:11307
Change-Id: I2ea33fd12f590b6031ac85f75708f6cc88a266b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162902
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30423}
This CL adds the missing support for multi-channel in the code that
provides an optional and configurable delay to be added to the
microphone signal.
The CL also makes the creation of the delay object conditional on the
need for that support (this is important since this adds a significant
heap memory footprint)
Bug: webrtc:11314,chromium:1045910
Change-Id: I92d577e31af830945fe9d5ca2032000aad4266be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167525
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30392}
This is a reland of 48148dc840
Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
>
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
>
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}
Bug: webrtc:11278, webrtc:11295
Change-Id: I8e6f11457e283c83cae5581adcacdc4d3b5431bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167360
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30372}
This reverts commit 48148dc840.
Reason for revert: Causing tests to timeout, see bugs.webrtc.org/11295
Original change's description:
> Change log level of AEC3 buffer info to VERBOSE
>
> Otherwise, test logs become very verbose:
> https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
> See linked issue.
>
> Bug: webrtc:11278
> Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30233}
TBR=saza@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11278
Change-Id: I283648a6d4d58cfe7af7a646d915122207883007
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167180
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30357}
This CL reduces the complexity of the Subtractor.ConvergenceMultiChannel
test by
1. Slightly reducing the amount of tested combinations for the non-debug
mode.
2. Drastically reduce the amount of tested combinations for the debug
mode.
Bug: webrtc:11295
Change-Id: I56bfa4a1463d26e5217b6a4d7f2ef54de7aab512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166529
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30343}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This is a reland of f3aa6326b8
Original change's description:
> Replace the ExperimentalAgc config with the new config format
>
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
>
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
>
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}
Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
The test is likely timing out on iOS simulator (see bug). Maybe I'm
going a bit overboard here :) if you want to keep all the cases I
removed, you can run some cases in one test method and the others in
another test method. Are the cases I removed particularly important?
Bug: webrtc:11284
Change-Id: I8f2e8830f931594c3471d1c20a2654e258b9fcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166169
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30277}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
This reverts commit f3aa6326b8.
Reason for revert: Breaks downstream project.
Original change's description:
> Replace the ExperimentalAgc config with the new config format
>
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
>
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
>
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}
TBR=saza@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:5298
Change-Id: I794d2ab4b8caa5330c5ad490ba604646a249a1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164530
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30153}
This CL replaces the use of the ExperimentalAgc config with
using the new config format.
Beyond that, some further changes were made to how the analog
and digital AGCs are initialized/called. While these can be
made in a separate CL, I believe the code changes becomes more
clear by bundling those with the replacement of the
ExperimentalAgc config.
TBR: saza@webrtc.org
Bug: webrtc:5298
Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30149}
The two functions have a lot of shared logic and locking. This CL consolidates that into a single function.
Bug: webrtc:111235
Change-Id: Ib1c32165dbf0e212c7d4b0753bcbb5ffd05eb6fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163022
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30144}
This CL ensures that the pre-amplifier and the high-pass filter
submodules are not reallocated more than needed.
Bug: webrtc:5298
Change-Id: I7ed23807d4d2d9fef0eda2e7dca9de9b0b1a4649
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163988
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30143}
This brings the two ProcessStream functions closer in implementation.
Additionally, the error checking that is currently done in the period of not holding the lock seems cheaper than releasing and reacquiring the capture lock.
Bug: webrtc:11235
Change-Id: Ib4afc68afb419fcabbb8cf08a3a2e61d2c12acda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30140}
