Commit graph

1057 commits

Author SHA1 Message Date
Markus Handell
97c4458c8f PlatformThread: add support for detached threads.
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}
2021-04-21 07:44:31 +00:00
Gustaf Ullberg
437d129ef5 AEC3: Avoid overcompensating for render onsets during dominant nearend
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}
2021-04-20 12:33:02 +00:00
Alessio Bazzica
980c4601e1 AGC2: retuning and large refactoring
- 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}
2021-04-14 19:01:01 +00:00
Alessio Bazzica
61982a7f2d AGC2 lightweight noise floor estimator
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}
2021-04-14 15:56:41 +00:00
Per Åhgren
dea5721efb Adding g3doc for AudioProcessingModule (APM)
Bug: webrtc:12569
Change-Id: I8fa896a5afa9791ad6d8c2b5011d1e75ca068df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215141
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33726}
2021-04-14 09:40:25 +00:00
Gustaf Ullberg
1fad94f502 Remove ErleUncertainty
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}
2021-04-14 07:01:14 +00:00
Alessio Bazzica
11bd143974 AGC2 add an interface for the noise level estimator
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}
2021-04-08 07:34:22 +00:00
Alessio Bazzica
70b775d77f AGC2 noise estimator code style improvements
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}
2021-04-07 11:57:55 +00:00
Alessio Bazzica
8aaa604375 AGC2 new data dumps
Bug: webrtc:7494
Change-Id: Id288dd426e1c2754805bc548fbffe0eaeaacf3da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213420
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33605}
2021-03-31 14:55:42 +00:00
Alessio Bazzica
841d74ea80 AGC2 periodically reset VAD state
Bug: webrtc:7494
Change-Id: I880ef3991ade4e429ccde843571f069ede149c0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213342
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33604}
2021-03-31 14:15:10 +00:00
Alessio Bazzica
b995bb86df AGC2 size_t -> int
Bug: webrtc:7494
Change-Id: I5ecf242e83b509931c1764a37339d11506c5afc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213341
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33600}
2021-03-31 11:18:30 +00:00
Per Åhgren
19775cbd29 Reland "Reduce complexity in the APM pipeline when the output is not used"
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}
2021-03-16 09:33:02 +00:00
Per Åhgren
15179a9986 Allowing reduced computations in the noise suppressor when the output is not used
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}
2021-03-16 09:28:42 +00:00
Per Åhgren
8ee1ec82e4 Allowing reduced computations in the AEC3 when the output is not used
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}
2021-03-16 09:16:32 +00:00
Per Åhgren
db5d728878 Add refined handling of the internal scaling of the audio in APM
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}
2021-03-15 19:12:02 +00:00
Gustaf Ullberg
fdd6099348 Rework transient suppressor configuration in audioproc_f
The transient suppressor can be configured as:
0 - Deactivated
1 - Activated with key events from aecdump
2 - Activated with continuous key events (for debugging purposes)

Bug: webrtc:5298
Change-Id: I116eb08ad50178dc5116d5d967084e6c9967f258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211869
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33464}
2021-03-15 15:19:09 +00:00
Per Åhgren
55bc077b45 Add one frame (10 ms) of silence in APM output after unmuting
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}
2021-03-13 01:05:45 +00:00
Ilya Nikolaevskiy
8097935df3 Revert "Reduce complexity in the APM pipeline when the output is not used"
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}
2021-03-11 11:41:20 +00:00
Per Åhgren
aa6adffba3 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}
2021-03-11 10:06:58 +00:00
Per Åhgren
34fdc92119 Add audioproc_f support for testing the runtime settings of whether the output is used
Bug: b/177830919
Change-Id: Iddcb79000f471eac165e3f44f14fad41435e6ccb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211241
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33426}
2021-03-10 23:19:30 +00:00
Per Åhgren
652ada5029 Enabling a safe fall-back functionality for overruns in the runtime settings
Bug: b/177830919
Change-Id: I9369f6fc004ceb2b626d33b36262bc8aeabdb1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206988
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33371}
2021-03-03 12:06:54 +00:00
Gustaf Ullberg
09226fc832 Disable high-pass filtering of the AEC reference
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}
2021-02-23 07:06:11 +00:00
Danil Chapovalov
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
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}
2021-02-22 12:53:23 +00:00
Per Åhgren
8eea117dea Make PostRuntimeSetting pure virtual
Bug: b/177830919
Change-Id: I92e30e9b65c8f851444268f0824a676044504814
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206640
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33252}
2021-02-12 22:49:43 +00:00
Alessio Bazzica
5a585952da Update AGC2 tuning
Bug: webrtc:7494
Change-Id: Ifcc5b6c846476ce7d6862fba2cb53e426b5855dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206800
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33238}
2021-02-11 18:14:34 +00:00
Per Åhgren
0a144a705a Adding initial support for lock-less informing of muting
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}
2021-02-09 12:08:54 +00:00
Per Åhgren
68c225d76d Make 48 kHz maximum rate default for all devices
Bug: b/169918549
Change-Id: I2f4b7ced5ae6efcae3cd59c0a42610a54f5e2dc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203260
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33192}
2021-02-08 15:06:05 +00:00
Per Åhgren
879d33b9f8 Add more refined control over dumping of data and the aecdump content
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}
2021-02-06 00:36:10 +00:00
Sam Zackrisson
53610223a8 Delete unused function webrtc::AudioProcessing::MutateConfig
Bug: None
Change-Id: Ibc70e5246a3f7b89775c65a19c808c1f030b8ac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205522
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33147}
2021-02-03 12:55:23 +00:00
Alessio Bazzica
6f75f6b3bd APM: add AGC2 SIMD kill switches in AudioProcessing::Config::ToString()
Bug: webrtc:7494
Change-Id: Icba5f6be689a57ef4748ae816565349fd1ad2108
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205322
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33146}
2021-02-03 12:31:56 +00:00
Andrey Logvin
e7c79fd3d6 Remove from chromium build targets that are not compatible with it.
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}
2021-02-01 13:46:19 +00:00
Andrey Logvin
b79acd8ce9 Format webrtc/modules/audio_processing/transient/BUILD.gn file
Bug: webrtc:12404
Change-Id: I3cb6b96255409709fa3144d2f83d13d12e39ab2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204064
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33107}
2021-01-29 16:33:19 +00:00
Per Åhgren
c2ae4c8a37 Allow separate dump sets for the data dumper in APM
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}
2021-01-20 16:38:17 +00:00
Gustaf Ullberg
5c3ff6b60f Switch to enable the HMM transparent mode classifier
Bug: chromium:1155071,webrtc:12265,chromium:1155477
Change-Id: I9d3119e9cbfdd5d7b41de2ed0f9dec92f7bf753d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202258
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33037}
2021-01-19 16:26:11 +00:00
Alessio Bazzica
42eef86c4f Remove unused code in APM
- 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}
2021-01-18 13:40:27 +00:00
Mirko Bonadei
e5f4c6b8d2 Reland "Refactor rtc_base build targets."
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}
2021-01-15 17:00:05 +00:00
Mirko Bonadei
7acc2d9fe3 Revert "Refactor rtc_base build targets."
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}
2021-01-14 21:27:38 +00:00
Hua, Chunbo
e61a40e4ef Fix typo in audio processing header.
Bug: None
Change-Id: Ia47f8229cdfbbcfec3aa7539c587d4e1fdc4ae3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200926
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32974}
2021-01-14 10:59:13 +00:00
Hua, Chunbo
dbcaff0925 Fix AudioProcessing::Config::ToString() implementation.
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}
2021-01-14 10:27:18 +00:00
Alessio Bazzica
76714a6cc8 AGC2 minor code clean up
Dead code removed plus const ref std::string to avoid copies.

Bug: webrtc:7494
Change-Id: Ic408a810ae310fea942f25fc697ab81017c8a739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201624
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32968}
2021-01-14 08:03:17 +00:00
Mirko Bonadei
69241a93fb 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}
2021-01-11 18:32:30 +00:00
Alessio Bazzica
5247070f5d RNN VAD: add missing CPU features to test FC and GRU layers
Bug: webrtc:10480
Change-Id: I6c49e728ed61647b098c20a6d8a856005066ab75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200840
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32930}
2021-01-10 12:42:39 +00:00
Alessio Bazzica
524f682184 SSE2 and NEON kill switches for AGC2
Bug: webrtc:7494
Change-Id: I221b717b5cf3c41b7b637e9234d1e339a0e6c7e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199967
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32918}
2021-01-07 11:41:28 +00:00
Alessio Bazzica
ed9f5f85fd RNN VAD optimizations: VectorMath::DotProduct() NEON arm64
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}
2020-12-30 10:47:01 +00:00
Jesús de Vicente Peña
16dd6b0ce4 Fixing the bounds for the max and min erle in the erle instantaneous estimation.
Bug: webrtc:12283
Change-Id: I534a5da30e575f9b100046fff31a84774afbf67f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196654
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32834}
2020-12-15 14:29:52 +00:00
Jesús de Vicente Peña
3bda2362f1 Erle metric: avoid the decreasing of the metric when there is no render activity.
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}
2020-12-09 11:48:11 +00:00
Alessio Bazzica
e8ee462681 RNN VAD: FC layer simplified
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}
2020-12-09 00:04:50 +00:00
Alessio Bazzica
4e9c5b592a RNN VAD: GRU layer optimized
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}
2020-12-08 15:37:38 +00:00
Gustaf Ullberg
992a96f68e AEC3: Prevent diverging coarse filter from influencing the refined filter
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}
2020-12-08 15:05:23 +00:00
Per Åhgren
12dc274749 Make the static variable kChunkSizeMs constexpr
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}
2020-12-08 10:46:15 +00:00