The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
Bug: webrtc:9878
Change-Id: Ic912d67455fcef4895566edb8fef62baf62d7cfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156440
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29454}
This change keeps the original 48 kHz signal and uses it for the
fullband processing given that the following requirements are
fulfilled:
- Input signal is 48 kHz
- Output signal is 48 kHz
- Multiband processing is performed at 32 kHz
- The multiband processing does not modify the original signal
This avoids unnecessary, lossy resampling and band merging.
Bug: b/130016532
Change-Id: I690c26faba07eab0cbff6c0a95a81d89255dd1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155966
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29425}
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 adds support in the echo subtractor for handling multiple
capture and render channels.
The changes have passed bitexactness tests for substantial set
of mono recordings.
Bug: webrtc:10913
Change-Id: Ib448c9edf172ebc31e8c28db7b2f2a389a53adb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155168
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29389}
Currently, APM fakes multichannel in two ways:
- With injected AECs, capture processing is only performed on the left
channel. The result is copied into the other channels.
- With multichannel render audio, all channels are mixed into one
before analysing.
This CL adds a flag to disable these behaviors, ensuring proper
multichannel processing happens throughout the APM pipeline.
Adds killswitches to separately disable render / capture multichannel.
Additionally - AEC3 currently crashes when running with multichannel.
This CL adds the missing pieces to at least have it run without
triggering any DCHECKS, including making the high pass filter properly
handle multichannel.
Bug: webrtc:10913, webrtc:10907
Change-Id: I38795bf8f312b959fcc816a056fba2c68d4e424d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152483
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29248}
This CL allows the user to have more refined control over what band
splitting-scheme is used inside the audio processing module.
Bug: webrtc:6181
Change-Id: I236c3b1f96ab80cc4ffb8c39c045c034764567a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29189}
In this CL:
- Render signal analyzer considers a frequency bin a narrow band
(peak) if any channel exhibits narrowband (-peak) behavior.
- The unit tests have to fill frames with noise because small
inaccuracies in the FFT spectrum lead to consistent "narrow bands"
despite spectrum being essentially flat.
Bug: webrtc:10913
Change-Id: I8fa181412c0ee1beeacfda37ffef18251d5f0cd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151912
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29176}
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 reverts commit 81c0cf287c.
Reason for revert: internal test failures
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}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
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}
This CL adds the following options:
pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file)
write the processed capture samples to a given vector
Bug: webrtc:10808
Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208
Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28826}
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
Add a PlayoutVolumeChange RuntimeSetting. Trigger an echo path change when the playout volume is changed.
Bug: webrtc:10608
Change-Id: I1e736b93c1865d08c7d2582f6fe00216c1e1f72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135746
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27913}
This CL extends the supported runtime settings in
APM to also comprise the AGC2 fixed gain.
The CL was originally created by Adam Whiteside.
Bug: webrtc:10574
Change-Id: I79b3d6501f1e202b66a9b6018f8a493a56b01f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27782}
The one with enum.IntFlag is not feasible. An attempt is done here:
https://webrtc-review.googlesource.com/c/src/+/133884
It requires re-writing QualityAssessment to Python3 which is too much
work for little benefit. (I tried, but couldn't get the unit-tests to
pass for both 2 and 3.)
The second one is not a real todo.
TBR=alessiob@webrtc.org
NOPRESUBMIT=True
Bug=None
NOTRY=True
Change-Id: Ia25817533cd504c30490f86e4058f0b2d59dd39c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133908
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27715}
This CL allows audioproc_f to overrule any runtime settings for the
pre-amplifier gain that are present in the aecdump file.
Bug: webrtc:10546
Change-Id: I74dbf8d043f59b516bf0abc80f266e965af0754d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132558
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27598}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This CL extends, and partly corrects, the benchmarking
code in audioproc_f to provide statistics for the API
call durations in audioproc_f
Bug: chromium:939791
Change-Id: I4c26c4bb3782335f13dd3e21e6f861842539ea62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129260
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27443}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This adds a flag to audioproc_f to generate a custom call order
file from an AEC dump. This file can be used to get more realism
when simulating with wav-files.
Bug: webrtc:10393
Change-Id: I245533d18affaab2f6cef53138332d7d83c71822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126782
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27104}
In this CL we avoid the propagation of the echo control factory to the AudioProcessing instance when this is not set. That propagation was unnecessarily overriding the echo control factory that might have been already set on that AudioProcessing instance.
Change-Id: Ife8f479bc7a81c35ecf656e7d0ddfcc98981c74f
Bug: webrtc:10344
Reviewed-on: https://webrtc-review.googlesource.com/c/123765
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26802}
This Config configuration will eventually replace the AudioProcessing::noise_suppression() interface.
This also introduces a proxy NoiseSuppression, returned by AudioProcessing::noise_suppression.
Without this proxy, ApplyConfig could overwrite NS settings for clients who currently use noise_suppression(). For example, the following code will not preserve the noise suppression level:
apm->noise_suppression()->set_level(NoiseSuppression::kHigh);
auto cfg = apm->GetConfig();
apm->ApplyConfig(cfg);
The NoiseSuppression instance returned by noise_suppression() has no way to update the config inside APM, so GetConfig() will return an out-of-date config which is then re-applied. This CL adds a proxy that makes this update, by forwarding Enable() and set_level() calls to ApplyConfig().
Drive-by change: AudioProcessing::Config substructs are reordered to mirror the capture processing pipeline.
Tested: Ran ToT and this CL builds of audioproc_f and verified identical settings/aecdumps.
Bug: webrtc:9947
Change-Id: I823eade894be115c254d656562564108b2b63b1f
Reviewed-on: https://webrtc-review.googlesource.com/c/116521
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26248}
This CL adds two unit tests to make sure that, when an echo path gain
change occurs, the echo canceller is notified.
Such a change can be caused by (i) a pre-amplifier gain change or
(ii) an analog gain change.
Bug: webrtc:7494
Change-Id: Ia47cfbbc5694340cd3e760d8d3c3393f79897a9d
Reviewed-on: https://webrtc-review.googlesource.com/c/111780
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26190}
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}
In this CL a more precise estimation of the Erle is introduced. This is done by creating different estimators that are specialized in different regions of the linear filter. An estimation of which regions were used for generating the current echo estimate is performed and used for selecting the right Erle estimator.
Bug: webrtc:9961
Change-Id: Iba6eb24596c067c3c66d40df590be379d3e1bb7b
Reviewed-on: https://webrtc-review.googlesource.com/c/109400
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25707}
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).
Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.
Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
Additionally, AudioProcessing::GetStatistics(bool) is made pure
virtual and the default implementation in AudioProcessing is removed.
Deprecation PSA:
https://groups.google.com/forum/#!msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ
Bug: webrtc:9947, webrtc:8572
Change-Id: I123402bf7d6c49f3613154c469b818109d8fad43
Reviewed-on: https://webrtc-review.googlesource.com/c/108783
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25463}
This CL creates a new API for the parser of APM json config that
that provides an explicit way for the user to know when there has
been an issue in the parsing of the json config data.
Bug: webrtc:9921
Change-Id: Idd8f40529f40ab6871efb5b356c0fd2cea21b7d9
Reviewed-on: https://webrtc-review.googlesource.com/c/107841
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25355}
This CL modifies the internal data logging and the audioproc_f tool
to allow controlling that via the command line, rather than solely via a
build flag. The logging of internal data is by default off.
Bug: webrtc:5298
Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107713
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25352}
This is a reland of 5ccdc1331f
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
This reverts commit 5ccdc1331f.
Reason for revert: Breaks downstream project.
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ia79cd6066ecfd1511c34f1b30fd423e560ed6854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9884
Reviewed-on: https://webrtc-review.googlesource.com/c/107160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25276}
Macros defined in rtc_base/flags.h are intended to be used to define
flags in WebRTC's binaries (e.g. tests).
They are currently not prefixed and this could cause problems with
downstream clients since these names are quite common.
This CL adds the 'WEBRTC_' prefix to them.
Generated with:
for x in DECLARE DEFINE; do
for y in bool int float string FLAG; do
git grep -l "\b$x\_$y\b" | \
xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
done
done
git cl format
Bug: webrtc:9884
Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
Reviewed-on: https://webrtc-review.googlesource.com/c/106682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25270}
This is a new attempt to reduce the filter divergence
during double-talk without regressing in clock-drift
scenarios.
- The error_floor in decreased to allow for slow adaptation
when the filter performs well.
- The leakage_diverged is increased to allow for fast adaptation
when the shadow filter performs better.
- A new parameter, error_ceil, was added to stop the filter from
adapting too fast.
Bug: webrtc:9746,chromium:883264
Change-Id: Ie2868d2388b48412a192a004ec13f9eff34517b8
Reviewed-on: https://webrtc-review.googlesource.com/c/100460
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25063}
This CL adds the ability to toggle the AGC2 adaptive digital mode in
audioproc_f
Bug: webrtc:5298
Change-Id: If1567d8c87f88992dff89253edb293a56cee0a73
Reviewed-on: https://webrtc-review.googlesource.com/c/103361
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24954}
This is a reland of 2fbb83b16b
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
Bug: webrtc:9535
Change-Id: I172706c6729cac4eb6afde1ebd6fc8f3a289d6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/102881
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24943}
This change does three things:
- Move rtc_json into rtc_base/strings/, a non-API directory more fitting to
its purpose.
- Make a target for the currently unused json_unittest.
- Make the code available for use in non-test code again.
Bug: webrtc:9802
Change-Id: Id964a8a4b47b732a962a364894a4dbd3e7f4650f
Reviewed-on: https://webrtc-review.googlesource.com/103126
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24932}
This reverts commit 2fbb83b16b.
Reason for revert: Speculative revert over failing Chromium bot:
https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28M%20Nexus5X%29/117
Original change's description:
> Remove APM-internal usage of EchoControlMobile
>
> This is a sibling CL to a similar one for EchoCancellation:
> https://webrtc-review.googlesource.com/c/src/+/97603
>
> - EchoControlMobileImpl will no longer inherit EchoControlMobile.
> - Removes usage of AudioProcessing::echo_control_mobile() inside most of
> the audio processing module and unit tests.
>
> The CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (comfort noise, routing mode), but prints an
> error message when unsupported settings are encountered.
>
> Tested: audioproc_f with .wav and aecdump inputs.
> Bug: webrtc:9535
> Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
> Reviewed-on: https://webrtc-review.googlesource.com/101621
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24888}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I1f8a27ac291f2cdc16c8daa32e399b74d489dbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/102642
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24895}
This is a sibling CL to a similar one for EchoCancellation:
https://webrtc-review.googlesource.com/c/src/+/97603
- EchoControlMobileImpl will no longer inherit EchoControlMobile.
- Removes usage of AudioProcessing::echo_control_mobile() inside most of
the audio processing module and unit tests.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (comfort noise, routing mode), but prints an
error message when unsupported settings are encountered.
Tested: audioproc_f with .wav and aecdump inputs.
Bug: webrtc:9535
Change-Id: I63c3c81bcaf44021315978e1a0f3e42173b988ce
Reviewed-on: https://webrtc-review.googlesource.com/101621
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24888}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
Original CL:
https://webrtc-review.googlesource.com/c/src/+/97603
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
The CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Revert CL:
https://webrtc-review.googlesource.com/c/src/+/100305
Bug: webrtc:9535
TBR: gustaf@webrtc.org
Change-Id: I9248046b3a6a82df6221e502481836948643a991
Reviewed-on: https://webrtc-review.googlesource.com/100461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24749}
This CL:
-Adds the option to log the aec3 parameters used for a simulation.
-Cleans up the logging of the custom setting of aec3 parameters to
instead rely on the newly added logging.
Bug: webrtc:8671
Change-Id: If73a73d08e5a5077416033ded598a83fb1ade3e0
Reviewed-on: https://webrtc-review.googlesource.com/100381
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24742}
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}
This reverts commit 1a03960e63.
Reason for revert: breaks downstream projects.
Original change's description:
> Remove APM internal usage of EchoCancellation
>
> This CL:
> - Changes EchoCancellationImpl to inherit privately from
> EchoCancellation.
> - Removes usage of AudioProcessing::echo_cancellation() inside most of
> the audio processing module and unit tests.
> - Default-enables metrics collection in AEC2.
>
> This CL breaks audioproc_f backwards compatibility: It can no longer
> use all recorded settings (drift compensation, suppression level), but
> prints an error message when such settings are encountered.
>
> Some code in audio_processing_unittest.cc still uses the old interface.
> I'll handle that in a separate change, as it is not as straightforward
> to preserve coverage.
>
> Bug: webrtc:9535
> Change-Id: Ia4d4b8d117ccbe516e5345c15d37298418590686
> Reviewed-on: https://webrtc-review.googlesource.com/97603
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24724}
TBR=gustaf@webrtc.org,saza@webrtc.org
Change-Id: Ifdc4235f9c5ee8a8a5d32cc8e1dda0853b941693
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/100305
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24729}
This CL:
- Changes EchoCancellationImpl to inherit privately from
EchoCancellation.
- Removes usage of AudioProcessing::echo_cancellation() inside most of
the audio processing module and unit tests.
- Default-enables metrics collection in AEC2.
This CL breaks audioproc_f backwards compatibility: It can no longer
use all recorded settings (drift compensation, suppression level), but
prints an error message when such settings are encountered.
Some code in audio_processing_unittest.cc still uses the old interface.
I'll handle that in a separate change, as it is not as straightforward
to preserve coverage.
Bug: webrtc:9535
Change-Id: Ia4d4b8d117ccbe516e5345c15d37298418590686
Reviewed-on: https://webrtc-review.googlesource.com/97603
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24724}
Added back the 'agc2 level estimation' flag. Also added a flag for
moving the level measurement before AEC and NS. This is to run offline
experiments with audioproc_f.
Bug: webrtc:7494
Change-Id: I3e3ffceede7166b754130be2b707b620ba527e9f
Reviewed-on: https://webrtc-review.googlesource.com/97442
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24657}
Also read and apply settings when parsing and replaying dumps.
The implementation contains
* an extra field in debug.proto for the runtime settings
* code in AudioProcessingImpl to initiate the logging of the RS to the
AecDump
* code in aec_dump/ to log the RS in the AecDump
* code in test/ for re-playing the RS. E.g. for APM simulation with
audioproc_f.
Bug: webrtc:9138
Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f
Reviewed-on: https://webrtc-review.googlesource.com/70502
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24647}
This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.
To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.
Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This was done by
* adding an EchoMetric class to EvaluationScore
* passing an echo metric binary path from the cmd arguments to the
EvaluationScoreWorkerFactory
* passing the render input filepath to the Evaluator.
The echo score is supposed to be computed by the provided binary. It
should print the echo score in [0.0, 1.0] to a text file. It should
satisfy the cmd flags in its invocation in EchoMetric._Run()
Bug: webrtc:7494
Change-Id: I397013d6ed17659ea01d0623d98a14d4fcdcc161
Reviewed-on: https://webrtc-review.googlesource.com/97022
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24537}
This CL introduces the ability to control the usage of the shadow filter
output in the echo canceller output.
Bug: webrtc:9694,chromium:879451
Change-Id: I01f90de60de1799b32892051c176bda5e1a8d33e
Reviewed-on: https://webrtc-review.googlesource.com/97020
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24506}
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).
Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
During this work a parameter is added to the configuration file for the AEC3 that allows to enable or disable the use of a different ERLE estimation for the render onsets.
Bug: webrtc:9677
Change-Id: I467f2cd20683fee06b69c0ba51a90816c9e14f29
Reviewed-on: https://webrtc-review.googlesource.com/96082
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24470}
This CL adds adaptive handling of platforms where the echo path has
a strong gain above 10 kHz. A configurable offset is adaptively applied
depending on the amount of echo and mode of the echo suppressor.
Bug: webrtc:9663
Change-Id: I27dde6dc23b04a76a3be8c49d7fc9e226b9137e6
Reviewed-on: https://webrtc-review.googlesource.com/95947
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24448}
This CL allows selecting an echo suppressor behavior which is specific
for whether the nearend is dominant, or the echo is dominant.
The changes in this CL are bitexact.
Bug: webrtc:9660
Change-Id: Ie32e65efe47e692de6d6a22a7ad3b469d745fd6b
Reviewed-on: https://webrtc-review.googlesource.com/95725
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24434}
This CL removes the UpdateGainIncrease code that is not used anymore.
The CL has been tested for bit exactness.
Bug: webrtc:8671
Change-Id: I4fcf26c3b4b5bba760ee139416ddefac86a36c2e
Reviewed-on: https://webrtc-review.googlesource.com/95940
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24425}
This CL adds a quiet mode for audioproc_f and hooks up the verbose
output of the AEC3 settings read from the JSON input file to that.
Bug: webrtc:8671
Change-Id: I93bbd1efc6502649da7b2b3e9f7557e9c184b0ed
Reviewed-on: https://webrtc-review.googlesource.com/95700
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24416}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I48b9deaad873aee597f56ebd33814420024e0d58
Reviewed-on: https://webrtc-review.googlesource.com/95645
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24405}
This CL removes the unused coherence computation from AEC3. This CL
only removes unused code, the output of AEC3 does not change.
Bug: webrtc:8671
Change-Id: Ie127c5ec64e29414f1e1570511d57a4d09fc9145
Reviewed-on: https://webrtc-review.googlesource.com/95650
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24403}
This CL adds functionality for applying an optional fixed delay in AEC3
to the capture signal
Bug: webrtc:9647
Change-Id: Id3b3f896bcf203e6611298dc804c3c80da9f1883
Reviewed-on: https://webrtc-review.googlesource.com/95142
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24399}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I0e1462fa6e35944f7dbb02580f1db09401c8f7c8
Reviewed-on: https://webrtc-review.googlesource.com/95484
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24394}
This CL:
-Adds another config parameter that controls the duration of the initial
state.
-Adds reading of that parameter in audioproc_f from the json settings file.
-Adds missing reading of another parameter in audioproc_f from the json
settings file.
Bug: webrtc:8671
Change-Id: Ie6164c360492de5e6b0ade8838bbabe214560b5e
Reviewed-on: https://webrtc-review.googlesource.com/94621
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24360}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I1be59a9277aad8f51765c26e34ab16b63bcaeb42
Reviewed-on: https://webrtc-review.googlesource.com/94774
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24340}
This allows clients to stop using the old pointer-to-submodule interfaces
for enabling/disabling AEC2 and AECM.
The legacy suppression level flag for AEC2 is not yet activated.
NoTry=TRUE
Bug: webrtc:9535
Change-Id: Ie2c3378d832a6b393aec656d96597f85e299f500
Reviewed-on: https://webrtc-review.googlesource.com/94771
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24328}
This CL adds a flag to optionally disable the digital gain control in
ExperimentalAgc. With the flag, Experimental Agc (henceforth AGC1)
only controls the adaptive analog gain. This flag can be combined to
that which activates AGC2. That way, one can enable the hybrid AGC
configuration AGC1 analog only + AGC2 fixed+adaptive digital.
Previously, there was a flag "use_agc2_digital_adaptive" in
AgcManagerDirect. Our ambition was that to activate the hybrid mode
described above with this flag. The behavior of the flag was not
implemented.
To activate the hybrid mode after this CL, set
ExperimentalAgc::digital_adaptive_disabled=true and
AudioProcessing::Config::GainController2::enabled=true.
We also add flags for these settings in audioproc_f.
Then the required settings are currently
audioproc_f --agc2 1 --agc 1 --experimental_agc 1 \
--experimental_agc_disable_digital_adaptive 1 \
-i [INPUT]
Bug: webrtc:7494
Change-Id: Iea798dc3899cec83d30ba71caba787262fcaef41
Reviewed-on: https://webrtc-review.googlesource.com/89740
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24249}
Agc2 applies a digital gain to the nearend signal.
When the analog level changes, the digital gain calculation is no
longer valid. Therefore Agc2 should be notified to analog gain
changes.
This CL also allow audioproc_f to chain AGC1 and AGC2. In a dependent
CL we will allow using AGC1 for analog gain and AGC2 for digital
gain.
Bug: webrtc:7494
Change-Id: Id75b3728fbf2de1d84b7fba005e4670c7a2985d9
Reviewed-on: https://webrtc-review.googlesource.com/89387
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24231}
This reverts commit 771b50ca0b.
Reason for revert: Introduces error-prone config.
Original change's description:
> Add one-stop-shop for built-in AEC toggling in APM
>
> This does not change what AEC functionality is available.
> However, a client that only uses this interface - and not the submodule
> pointer accessors - gets simpler code, and is guaranteed not to run any
> two AECs in tandem.
>
> The submodule interface EchoControlMobile is being deprecated in
> https://webrtc-review.googlesource.com/c/src/+/89392
>
> Bug: webrtc:9535
> Change-Id: Id9326074e566be6d8768010fc421c457beff402c
> Reviewed-on: https://webrtc-review.googlesource.com/89386
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24066}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I43283a1b22538a4caa77313499989146b2ce67f1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/90060
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24067}
This does not change what AEC functionality is available.
However, a client that only uses this interface - and not the submodule
pointer accessors - gets simpler code, and is guaranteed not to run any
two AECs in tandem.
The submodule interface EchoControlMobile is being deprecated in
https://webrtc-review.googlesource.com/c/src/+/89392
Bug: webrtc:9535
Change-Id: Id9326074e566be6d8768010fc421c457beff402c
Reviewed-on: https://webrtc-review.googlesource.com/89386
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24066}
This CL adds two flags to audioproc_f. The flags control
AgcManagerDirect. The flags are
'--experimental_agc_agc2_level_estimator' and
'--experimental_agc_agc2_digital_adaptive'.
After this CL, the flags are be applied to AgcManagerDirect. The flags
have no effect in release-mode. They cause a crash in debug-mode.
In an upcoming CL, '--experimental_agc_agc2_level_estimator 1' will
replace the speech level estimation in ExperimentalAgc with that of
AGC2.
'--experimental_agc_agc2_digital_adaptive 1' will replace the digital
gain selection and application with that of AGC2.
These audioproc_f will activate both new settings:
./out/Target/audioproc_f --agc 1 --experimental_agc 1
--experimental_agc_agc2_digital_adaptive 1
--experimental_agc_agc2_level_estimator 1 --simulate_mic_gain 1
--simulated_mic_kind 2
See also https://webrtc-review.googlesource.com/c/src/+/79360
Bug: webrtc:7494
Change-Id: If0e65893dffdddb312e553787b8cedaf9a334323
Reviewed-on: https://webrtc-review.googlesource.com/86548
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23802}
The AGC submodule of APM changes analog gain. These gain changes are
typically ignored by the test tool audioproc_f.
There is an option of the test tool to take action on the gain
changes. It's the '--simulate_mic_gain' option. The option converts
the analog gain to a digital gain. The digital gain is applied to the
capture stream.
This change adds a new simulated microphone kind. The new microphone
has a gain curve defined by
modules/audio_processing/agc/gain_map_internal.h. That gain curve
defines how AGC1 expects a microphone to behave.
Bug: webrtc:7494
Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780
Reviewed-on: https://webrtc-review.googlesource.com/86128
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23801}
This CL removes the remaining beamformer parts from the APM.
Bug: webrtc:9402
Change-Id: I9ab2795bd2813d17166ed0925125257b82d98a74
Reviewed-on: https://webrtc-review.googlesource.com/83340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23694}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_processing'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
This removes beamformer references from audioproc_f, some non-beamformer tests, and a few other bits and bobs.
The beamformer is, after this, very decoupled from the remaining APM code.
Bug: webrtc:9402
Change-Id: Iaafc95517013d7a17723ef2329f17b5e09069bc9
Reviewed-on: https://webrtc-review.googlesource.com/83983
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23649}
In this work we introduce some changes on how the reverberation model for AEC3 is applied. Currently, the exponential modelling of the tails is applied over the linear echo estimates. That might result in an overestimation of the reverberation tails under certain conditions. In this work, the reverberation model is instead applied over an estimate of the energies at the tails of the linear estimate.
Additionally, the stationary estimator is changed so it does not disable the aec immediately after a burst of activity.
Bug: webrtc:9384,webrtc:9400,chromium:852257
Change-Id: Ia486694ed326cfe231fc688877c0b9b6e2c450ff
Reviewed-on: https://webrtc-review.googlesource.com/82161
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23599}
In this work, we change the behavior of the gain limiter so it also looks at the energy
on farend around the default delay for deciding the suppression gain
that should be applied at the initial portion of the call.
Bug: webrtc:9311,chromium:846724
Change-Id: I0b777cedbbd7fd689e72070f72237296ce120d3c
Reviewed-on: https://webrtc-review.googlesource.com/78960
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23400}
This CL is created from a work initiated at https://webrtc-review.googlesource.com/c/src/+/61160
The purpose of this work is to improve the performance of the echo canceler (AEC3) when the farend signal contains stationary noises:
- An stationarity estimator of the farend signal has been added for detecting the portions of the farend signal that are pure noise.
- When the echo canceler deals with a portion of the signal that contains basically noise, the echo suppressor is able to back-off and avoid the fading of the nearend speech.
Change-Id: Id4b87fc59f4765bf1fca36d1cab39a49aabe104a
Bug: webrtc:9193,chromium:836790
Reviewed-on: https://webrtc-review.googlesource.com/64141
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23024}
This CL increases the transparency in AEC3 during regions of low level
echo. What is done is:
-Low-level echoes are smoothly weighted so as to be deemed less
disturbing.
-The time-domain masking effect of the nearend speech is increased for
all frequencies.
-A separate, even more increased, time-domain masking effect is
introduced for lower frequencies.
-The intra-band masking is reduced to reduce the risk of echo leakage.
-The limiting of maximum gain due to filter-bank dynamics is removed
as the usecase for it could no longer be identified.
Bug: webrtc:9159,cromium:833801
Change-Id: I289b92919763124d6c5e5ede19e9a5917877c654
Reviewed-on: https://webrtc-review.googlesource.com/70421
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22915}
Add configuration fields for the pre-amplifier in the Audio Processing
Module. Also add flags and settings for the pre-amplifier in
audioproc_f.
Also make the setting stored in Aec Dumps. And make the setting
applied when playing back Aec Dumps in audioproc_f.
Bug: webrtc:9138
Change-Id: I4e59df200e1ebc56f06fae74ebf17d85858958a3
Reviewed-on: https://webrtc-review.googlesource.com/69560
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22876}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.
Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
This CL corrects some errors that were included in the CL for reading
the AEC3 options in the audioproc_f tool
Bug: webrtc:8671
Change-Id: Iecaee0ebf08f8a8f75aba1d395dd467a41b876f3
Reviewed-on: https://webrtc-review.googlesource.com/60870
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22384}
This refactoring makes it easier to experiment with injectable components.
Bug: webrtc:8732
Change-Id: I2cd2a8ff80516a76aec814af02b61778915f2217
Reviewed-on: https://webrtc-review.googlesource.com/60863
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22372}
This is a reland of 6f37ed78d9
CQ dry run OK except for missing iOS swarming bots.
NOTRY=True
Original change's description:
> Deprecate the adaptive level controller
>
> Level control handled by default-on AGC.
>
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> 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@{#22305}
Bug: none
Change-Id: I0b9b8e2f3457d5efd4603efbfbbc6b84651df315
Reviewed-on: https://webrtc-review.googlesource.com/60720
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22352}
This CL extends the options for the audioproc_f tool to match the options
for AEC3.
Bug: webrtc:8671
Change-Id: I39972eae33dba461b94118ec47a8560eb9cfe5a6
Reviewed-on: https://webrtc-review.googlesource.com/43120
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22344}
The AEC3 factory is now part of the WebRTC API.
Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
The functions replace some existing code and will be used in the
the new AutomaticGainController.
Bug: webrtc:7949
Change-Id: I9a32132d4a4699a507b8548a2eac10972a2f3fd6
Reviewed-on: https://webrtc-review.googlesource.com/53141
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22045}
WHAT: made a BUILD.gn with library and tests in the Audio Processing
Module Voice Activity Detector directory. Updated depending
code. Fixed a Clang warning.
WHY: to make it possible for a target to depend on just the VAD and
not the whole APM. There are other benefits:
* Sometimes faster compilation.
* The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making
a peerconnection shared object file without the VAD has to be done in
steps. The first step is a custom target for the VAD. Hence this Cl.
Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af
BUG: webrtc:5716, webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/47521
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21893}
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
The conversational_speech_generator tool now adjusts the level of
different speech segments.
Implementation:
The Turn and MultiEndCall::SpeakingTurn structs have an extra 'gain'
member. It's read and parsed in timing.cc and put in a Turn
struct. It's put in a SpeakingTurn struct in multiend_call.cc and read
and applied to the signal in simulator.cc
Bug: webrtc:7494
Change-Id: I9b82a896eb616c8b5ef14d41dfdfd085ef1d3fbb
Reviewed-on: https://webrtc-review.googlesource.com/26280
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21714}
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.
Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
This CL adds a way to insert a custom render-side pre-processor to
APM. The pre-processor operates in full-band mode before anything
else. Currently the render processing chain is (if everything is
enabled):
Network --> [Pre processing] --> [Band split] -->
[IntelligibilityEnhancer] --> [Echo canceller (read-only)] -->
[Band merge] --> Playout
Since the render pre processor and capture post processor have the
same interface, I renamed webrtc::PostProcessing into
webrtc::CustomProcessing.
The old APM factory method PostProcessing will be deprecated and
dependencies updated as part of webrtc:8665
NOTRY=True
Bug: webrtc:8665
Change-Id: Ia381cbf12e336d6587406a14d77243d931f69a31
Reviewed-on: https://webrtc-review.googlesource.com/29201
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21327}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
This tool is used downstream, so we want to christen rtc_tools as
a kind of api dir for tools. Tools in other locations should be
considered off limits.
I chose rtc_tools because video_quality_toolchain is already there,
which is also used downstream.
Bug: None
Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836
Reviewed-on: https://webrtc-review.googlesource.com/32340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21236}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8
Reviewed-on: https://webrtc-review.googlesource.com/28301
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21199}
This reverts commit 0af8370cb3.
Reason for revert: Breaks downstream
Original change's description:
> Fix all circular deps in audio_processing (but one).
>
> Arguably we should add a few more targets, for instance a utility
> target, but I tried to create as few targets as possible here based on
> the current usage.
>
> Bug: webrtc:6828
> Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a
> Reviewed-on: https://webrtc-review.googlesource.com/28020
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21025}
TBR=phoglund@webrtc.org,peah@webrtc.org
Change-Id: I423f027f6919cf4eb44b4e08c7cb38f0506ad0d7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/28940
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21027}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: If2740de2e4374eeae64b3d7599a52bb051594c6a
Reviewed-on: https://webrtc-review.googlesource.com/28020
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21025}
This is a minor change to generated Python code used for testing the echo likelihood metric.
Bug: webrtc:8573
Change-Id: Ifb2438fdd36c3ade8cd830df0d05917af0f77dec
Reviewed-on: https://webrtc-review.googlesource.com/26281
Commit-Queue: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20939}
The APM-QA tool produces clean-speech + noise + echo mixes with the
--additive_noise_tracks_path, --test_data_generators,
--echo_path_simulator flags. From this CL, it automatically produces
compressed Numpy annotations for the mixes. Annotations are placed in
the same folder as the mixes with name '${basename}-annotations.npz'.
TBR=alessiob@webrtc.org
NOTRY=True
Bug: webrtc:7494
Change-Id: I71941a4283594ef813de3ed65be31623bce5d7de
Reviewed-on: https://webrtc-review.googlesource.com/24960
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20844}
There is now an 'ExternalVad' class in the AnnotationsExtractor. The
Extractor takes an extra list of these in addition to the other
VADs. The external VAD runs an external program to generate the
annotations. Annotations are loaded and saved to a compressed Numpy format.
Also made a small fix to name a mixed file in a way so that files will not
be overwritten.
Also did some minor changes to the unittests.
TBR=alessiob@webrtc.org
Bug: webrtc:7494
Change-Id: I7816b04466be16cd635ac6ceab18cd7aad5325a4
Reviewed-on: https://webrtc-review.googlesource.com/23623
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20819}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=henrik.lundin@webrtc.org
Bug: None
Change-Id: I733a83f702fe11884d229a1713cfac952727bde8
Reviewed-on: https://webrtc-review.googlesource.com/23601
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20786}
Added possibility to extract audio_processing VAD annotations in the Quality Assessment tool.
Annotations are extracted into compressed Numpy 'annotations.npz' files.
Annotations contain information about VAD, speech level, speech probabilities etc.
TBR=alessiob@webrtc.org
Bug: webrtc:7494
Change-Id: I0e54bb67132ae4e180f89959b8bca3ea7f259458
Reviewed-on: https://webrtc-review.googlesource.com/17840
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20581}
Alternative VAD based on the existing one in WebRTC.
It is used to extract VAD annotations in APM-QA.
TBR=
Bug: webrtc:7494
Change-Id: I6af412742f804631ad4f3ba3ccf71a30d74de984
Reviewed-on: https://webrtc-review.googlesource.com/14553
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20404}
The struct containing the config for AEC3 is removed from
AudioProcessing::Config and is put in a new struct called
EchoCanceller3Config.
AEC3 should no longer be activated through
AudioProcessing::ApplyConfig. Instead an EchoCanceller3Factory
can be injected at AudioProcessing creation.
Bug: webrtc:8346
Change-Id: I27e3592e675eec3632a60c45d9e0d12514c2c567
Reviewed-on: https://webrtc-review.googlesource.com/11420
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20342}
