`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 CL moves the high-pass filter to run in the full-band domain
instead of the split-band domain.
Bug: webrtc:11193
Change-Id: Ie61f4a80afda11236ecbb1ad544bbd0350c7bbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161453
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30112}
Beyond making the digital AGC1 code properly support
multichannel, this CL also
-Removes deprecated debug logging code.
-Converts the gain application to be fully in floating point
which
--Is less computationally complex.
--Does not quantize the samples to 16 bit before applying the
gains.
Bug: webrtc:10859
Change-Id: I6020ba8ae7e311dfc93a72783a2bb68d935f90c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29886}
The changes have been shown to be bitexact on a large dataset.
Bug: webrtc:10859
Change-Id: Iedc0e9e944ebfabb717dd7fb4d2682c695da883e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159694
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29883}
This is a reland of 87a7b82520
Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
>
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
>
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
>
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
>
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}
Bug: webrtc:10895, b/143344262
Change-Id: I236f1e67bb0baa4e30908a4cf7a8a7bb55fbced3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158747
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29663}
This reverts commit 87a7b82520.
Reason for revert: Speculative revert. Breaks downstream projects.
Original change's description:
> Refactoring of the noise suppressor and adding true multichannel support
>
> This CL adds proper multichannel support to the noise suppressor.
> To accomplish that in a safe way, a full refactoring of the noise
> suppressor code has been done.
>
> Due to floating point precision, the changes made are not entirely
> bitexact. They are, however, very close to being bitexact.
>
> As a safety measure, the former noise suppressor code is preserved
> and a kill-switch is added to allow revering to the legacy noise
> suppressor in case issues arise.
>
> Bug: webrtc:10895, b/143344262
> Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29646}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I4d4025bda01f484979961fe57380a705e4d78397
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10895, b/143344262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158701
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29651}
This CL adds proper multichannel support to the noise suppressor.
To accomplish that in a safe way, a full refactoring of the noise
suppressor code has been done.
Due to floating point precision, the changes made are not entirely
bitexact. They are, however, very close to being bitexact.
As a safety measure, the former noise suppressor code is preserved
and a kill-switch is added to allow revering to the legacy noise
suppressor in case issues arise.
Bug: webrtc:10895, b/143344262
Change-Id: I0b071011b23265ac12e6d4b3956499d122286657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158407
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29646}
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
ApmTest.Process passes with unchanged reference files if
audio_processing_impl would initialize the VAD with
VoiceDetection::kLowLikelihood instead of kVeryLowLikelihood.
This was verified by testing this CL with that modification.
Bug: webrtc:9878
Change-Id: I4d08df37a07e5c72feeec02a07d6b9435f917d72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155445
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29395}
This CL removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.
Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
This is a reland of b7b8e30cb4
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
Bug: webrtc:10863
Change-Id: Ic626b99b099248f0d8a677dc4cfe1505e14ae7cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150330
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28949}
This reverts commit b7b8e30cb4.
Reason for revert: Broke ApmTest.Process test in internal iOS waterfall
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: Ia49e07b0c25c49da646917516e317f1d57cc4e84
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150326
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28948}
This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
of the audio processing module" which by mistake was reverted via a rebase in
another CL.
The CL changes the behavior of APM for 8 kHz so that it is internally
processed as 16 kHz.
Bug: webrtc:10863
Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28944}
This is a reland of 81c0cf287c
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
This CL changes the behavior of APM for 8 kHz so that it is internally
processed as 16 kHz.
Bug: webrtc:10863
Change-Id: Ie17de6551c6e984b60534820374a49ca298f06ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148800
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28929}
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.
The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.
Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
- add test that checks that the computed VAD probability is within
tolerance *1
- speed-up some tests by reducing the input length and skipping frames
- remove unused code in test_utils
- fix some comments
*1: RnnVadTest::RnnBitExactness is replaced by
RnnVadTest::RnnVadProbabilityWithinTolerance
Bug: webrtc:10480
Change-Id: I19332d06eacffbbe671bf7749ff4c92798bdc55c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133910
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27803}
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.
This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.
Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
This CL changes the APM unittests to use AEC3 instead of
AEC2.
Bug: webrtc:8671
Change-Id: I80f88dbafb7c31696abd8b7efb5a187a9fb30d1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129420
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27607}
This replaces the current usage of AudioProcessing::level_estimator()
in that test.
The unit tests that specifically test the level_estimator API are left
in place, until the level_estimator API itself is removed.
Bug: webrtc:9947
Change-Id: I73301c1478d2c9763bb49598a692142229102876
Reviewed-on: https://webrtc-review.googlesource.com/c/114550
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26049}
The test is refitted to use the AudioProcessingStats struct to get
reference data.
The old metrics do not map entirely injectively to the new ones, so the
reference protobuf and files are updated as well.
Bug: webrtc:9535
Change-Id: I546dca2979380e03895af0077bfc77ffd24abe36
Reviewed-on: https://webrtc-review.googlesource.com/100100
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24740}
All clients who do not own their own APM turn it off by default
(in WebrtcVoiceEngine). AECM with comfort noise is a little-exercised
code path. Configurability of this setting is going away, so we're
better off disabling it by default.
Bug: webrtc:9535
Change-Id: Iba839aa18e79ae29ff20bdf6e30de77870ba4143
Reviewed-on: https://webrtc-review.googlesource.com/89583
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24078}
This CL adds helper functions to be used for the spectral features
computation. Namely, it includes the following:
- band boundaries (frequency to FFT coeffcient index)
- band energy coefficients
- log band energy coefficients
- fixed size DCT table and computation
Bug: webrtc:9076
Change-Id: I03a8799b226d986bc1e37cefd0c3039f94b5592a
Reviewed-on: https://webrtc-review.googlesource.com/73687
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23170}
RNN implementation for the AGC2 VAD that includes a fully connected
layer and a gated recurrent unit layer.
Bug: webrtc:9076
Change-Id: Ibb8b0b4e9213f09eb9dbe118bbdc94d7e8e4f91b
Reviewed-on: https://webrtc-review.googlesource.com/72060
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23101}
Functions to estimate the inverse filter via LPC and compute the LP
residual applying the inverse filter.
This CL also includes test utilities, in particular BinaryFileReader,
used to read chunks of data and optionally cast them on the fly, and
Create*Reader() functions to read resource files available at test
time.
Bug: webrtc:9076
Change-Id: Ia4793b8ad6a63cb3089ed11ddad89d1aa0b840f6
Reviewed-on: https://webrtc-review.googlesource.com/70244
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22946}
This CL includes the patch from oprypin@webrtc.org, which is also applied
to the MIPS code (also affected), and the protobuf for ApmTest.Process
(audio_processing_unittest.cc), which used when WEBRTC_AUDIOPROC_FIXED_PROFILE
is set.
This change has been tested on mobile platforms.
Bug: webrtc:8200
Change-Id: Ic50a5ab57c16551397756b1fb473e1067b8e7ece
Reviewed-on: https://webrtc-review.googlesource.com/10811
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20394}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
This CL corrects the initialization of the AEC3, as well
as for the other submodules in the whole audio processing module
in the sense that it properly update the submodule states also
for the case when reinitialization is trigger from the render
side of the audio processing module.
Bug: chromium:736889,webrtc:7879
Change-Id: I423e963835d0c3227caa8e186b29031bcb912515
Reviewed-on: https://chromium-review.googlesource.com/549315
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18784}
Publicly available dataset of conversational speech audio recordings.
This CL includes the following:
- README.md: dataset description file, it also includes the scripts
- *.wav.sha1: hash files for each audio track in the dataset
The overall size of the wav files is ~36MB.
The primary intended use of this dataset is in combination with the conversational speech tool (see https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_processing/test/conversational_speech/), using which longer recordings with custom turn switch timing can be created.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2869833002
Cr-Commit-Position: refs/heads/master@{#18068}
Move the resources to //resources and upload them to Google Storage.
BUG=webrtc:6727
Review-Url: https://codereview.webrtc.org/2508943004
Cr-Commit-Position: refs/heads/master@{#15152}
