Commit graph

141 commits

Author SHA1 Message Date
Per Åhgren
cf4c872dbd APM: Make the GetStatistics call independent of the locks in APM
This CL changes the GetStatistics call in the audio processing module
(APM) to not aquire the render or capture locks in APM, while still
being thread-safe.
This change eliminates the risk of thread-priority inversion due to the
GetStatistics call.

Apart from the above the CL:
-Corrects the GetStatistics to not be const (it was const even though it
 aquired locks).
-Slightly changes the statistics reporting, so that the stats received
may be older than the most recent stats reported.

Bug: webrtc:11241
Change-Id: I00deb5507e004cbe6e4a19a8bad357491f86f4ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163982
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30131}
2020-01-02 15:45:14 +00:00
Per Åhgren
2e8e1c699e Open up for do the noise suppressor analysis on the linear AEC output
This CL allows the noise suppressor to use the linear AEC output
for analysis whenever that is available. This will potentially
lower the risk that the noise suppressor estimates the noise
based on echo.
The feature is off by default.

Bug: webrtc:5298,b:132164318
Change-Id: Idc6c8e197d96209d213819d87a8fb2533b7303ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162900
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30116}
2019-12-20 09:28:01 +00:00
Per Åhgren
c04242548c Make the high-pass filter operate in full-band
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}
2019-12-18 16:01:24 +00:00
Per Åhgren
ded86c1ad8 Remove remaining settings for using legacy AEC
This CL removes the remaining settings for using the legacy AEC.

It also adds a missing printout of the enforce_high_pass_filtering
parameter in the ToString method.

Bug: webrtc:11165
Change-Id: I58f0861bf1c6cd24bd83f4d3e394653b2fab3d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161683
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30050}
2019-12-10 10:31:07 +00:00
Per Åhgren
62ea0aaea0 Remove deprecated legacy AEC code
This CL removes the deprecated legacy AEC code.

Note that this CL should not be landed before the M80 release has been cut.

Bug: webrtc:11165
Change-Id: I59ee94526e62f702bb9fa9fa2d38c4e48f44753c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161238
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30036}
2019-12-09 10:37:49 +00:00
Yves Gerey
1fce3f8e55 Remove custom constructors for AudioProcessing::Config.
This CL follows the "Rule of zero".

Those constructors made no sense compared to default generated ones,
since all members are POD.
They were introduced to quiet a memory sanitizer warning,
which apparently was misleading.

As a bonus, the struct is now movable.

Bug: webrtc:11180, webrtc:9855
Change-Id: Iff9fd950bec8040bc6e7e7ece33cc49c5f453f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161381
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30023}
2019-12-06 06:49:04 +00:00
Per Åhgren
2512604705 Adding a copy constructor for the Config in AudioProcessing
Bug: webrtc:11180
Change-Id: I4621f83c0441fda55d0f81606174c004668dd6c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161325
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30015}
2019-12-05 13:58:20 +00:00
Per Åhgren
b810646c54 AEC3: Ensure that the high-pass filter is on whenever the AEC is active
This CL ensures that the high-pass filter is on whenever the echo
controller is on. This is important as the echo controller code assumes
that the external high-pass filter is active.

The CL also corrects the ToggleAec unit test (which started failing
after this code change).

Bug: webrtc:11159, chromium:1030179
Change-Id: Ief86eda8f7c67df1c25ac1a06d2cc0778e01196d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161228
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29998}
2019-12-04 09:38:04 +00:00
Per Åhgren
bcce4530d4 AEC3: Add flag for requiring a high pass filter effect before the AEC
Bug: webrtc:11159, chromium:1030179
Change-Id: Ic6b686a36efa433156a66252394fa551b8ed1837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161227
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29986}
2019-12-03 15:34:04 +00:00
Per Åhgren
b144c58973 Remove deprecated setting for activating multichannel processing
Bug: webrtc:10859
Change-Id: I86cae6a9b765bc807c00632ec7d743b754941f81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160780
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29966}
2019-12-02 07:47:53 +00:00
Per Åhgren
e14cb99408 Correct/update the activation of the multi-channel processing in APM
This CL removes the experimental status of the multi-channel processing
in APM, and accordingly updates the variable naming.

It also splits the activation of multi-channel processing to be separate
for render and capture.


Bug: webrtc:10859
Change-Id: I0e5d04dcb94b6637c33d97146231b8ddddbaea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29926}
2019-11-27 10:15:25 +00:00
Per Åhgren
c20a19cc4b Allow extracting the linear AEC output
This CL enables extracting the linear AEC output,
allowing for more straightforward
testing/development.

Bug: b/140823178
Change-Id: I14f7934008d87066b35500466cb6e6d96f811688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153672
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29789}
2019-11-13 11:33:53 +00:00
Mirko Bonadei
d4002a733d RTC_EXPORT missing symbols for Chromium's component build.
This CL adds a dependecy on rtc_base/system:rtc_export to rtc_event but
only when built as part of Chromium (since rtc::Event should not be
used outside of WebRTC).

It also adds other missing RTC_EXPORTS.

Bug: webrtc:9419
Change-Id: Ib338004a5404a6b3c7929e146c29ad42572632cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159692
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29781}
2019-11-12 21:50:01 +00:00
Per Åhgren
b8c1be5b6e Further AGC refactoring in preparation for adding multichannel support
Bug: webrtc:10859
Change-Id: If7d58a615a365a0b0f7b49e0cc2392b9bd5e2a0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159028
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29736}
2019-11-07 22:44:08 +00:00
Alessio Bazzica
7c19a706b0 Audio Processing Module: add play-out audio device runtime information
Add a runtime setting that notifies play-out audio device changes.
The payload is a pair indicating a device id and its maximum play-out
volume.

kPlayoutVolumeChange is now forwarded not only to capture, but also
render (required by render_pre_processor).

Bug: webrtc:10608
Change-Id: I8997c207422c1dcd1d53775397d6290939ef3db8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159002
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29725}
2019-11-07 13:33:09 +00:00
Per Åhgren
0cbb58e046 Reland "Refactoring of the noise suppressor and adding true multichannel support"
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}
2019-10-31 11:56:01 +00:00
Artem Titov
4778f6ce7a Revert "Refactoring of the noise suppressor and adding true multichannel support"
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}
2019-10-30 09:49:31 +00:00
Per Åhgren
87a7b82520 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}
2019-10-29 23:23:38 +00:00
Gustaf Ullberg
cb30726646 Remove deprecated Audio Processing APIs
This change removes the deprecated ChannelLayout versions of ProcessStream
and AnalyzeReverseStream.

Bug: webrtc:5298
Change-Id: I8a7e33e89cffac5eceecd00dfd3c96000643f51b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158529
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29641}
2019-10-29 10:07:08 +00:00
Gustaf Ullberg
8c51f2e9cd AnalyzeReverseStream with StreamConfig
Adding a version of AnalyzeReverseStream with audio parameters
described by StreamConfig. This is part of preparations for
multichannel APM in Chromium.

Bug: webrtc:10913
Change-Id: I7c4650eab8bd7fcdec970a7e4a8fa203f09bed9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157897
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29573}
2019-10-22 14:27:14 +00:00
saza
0bad15f2ed Remove the noise_suppression() pointer to submodule interface
Bug: webrtc:9878
Change-Id: I356afddb56cc1957e9d0415e2723f66e0e4ac522
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137517
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29499}
2019-10-16 11:55:15 +00:00
Sam Zackrisson
41478c7c1b Remove AudioProcessing::gain_control() getter
This change also resolves a bug in audioproc_f:
The implicit ApplyConfig calls to enable gain control settings in
aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig
call on line 500 using a config from line 292.

Compared to a ToT build including a fix for that bug, these changes
are bitexact on a large number of aecdumps.

[1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b

Bug: webrtc:9878
Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 09:23:16 +00:00
Mirko Bonadei
35214fcfe2 Add missing RTC_EXPORT for the component build.
Bug: webrtc:9419
Change-Id: I3225259fb4cc55e9820f590928795f4587f1e3cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29479}
2019-10-15 09:07:44 +00:00
saza
6787f232ae Remove AudioProcessing::level_estimator() getter
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}
2019-10-11 18:08:17 +00:00
Sam Zackrisson
0824c6f61a Delete voice_detection() pointer to submodule
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}
2019-10-07 13:06:05 +00:00
Sam Zackrisson
feee1e4c36 Add flag to APM to force multichannel even with AEC3
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}
2019-09-20 06:36:12 +00:00
Per Åhgren
fcbe4071ce Adding more refined control over choice of band-splitting
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}
2019-09-14 23:14:17 +00:00
Per Åhgren
c8626b6072 Reland "Reland Process 8 kHz audio as 16 kHz internally of the audio processing module"
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}
2019-08-23 14:24:48 +00:00
Artem Titarenko
7c4b0c56bf Revert "Reland Process 8 kHz audio as 16 kHz internally of the audio processing module"
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}
2019-08-23 13:11:13 +00:00
Per Åhgren
b7b8e30cb4 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}
2019-08-23 09:19:28 +00:00
Per Åhgren
d47941e018 Reland "Simplification and refactoring of the AudioBuffer code"
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}
2019-08-22 10:34:05 +00:00
Per Åhgren
1c602e39ce Process 8 kHz audio as 16 kHz internally of the audio processing module
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}
2019-08-21 14:21:53 +00:00
Artem Titov
59bbd65561 Add ToString method for AudioProcessing::Config
Bug: webrtc:10138
Change-Id: Icc0b40596867fd6fb8c1de5ca209dbafb504e50d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147726
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28745}
2019-08-02 10:01:56 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Niels Möller
825aad13dc Delete almost all includes of platform_file.h
Only remaining user is WavReader. Demote its constructor
accepting a PlatformFile to private, to refactor implementation
in a later cl.

Bug: webrtc:6463
Change-Id: I7b950be6f02073cb135dd0fab1190b9dc0de1fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144025
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28410}
2019-06-28 07:30:15 +00:00
Fredrik Hernqvist
ca362855e1 Add PlayoutVolumeChange RuntimeSetting.
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}
2019-05-10 14:12:23 +00:00
Per Åhgren
ea4c5df366 Remove the reporting of histogram data for AEC2
This CL removes the legacy reporting of histogram data for AEC2.

Bug: webrtc:5298
Change-Id: I838e729e0fb78d28e16de0fa79ddf5c857682d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135101
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27834}
2019-05-03 08:58:42 +00:00
Per Åhgren
6ee75fdfcb Allow setting the AGC2 fixed gain during runtime
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}
2019-04-26 10:05:45 +00:00
Sam Zackrisson
f0d1c03c31 Add replacement interface for webrtc::GainConrol
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
   to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
   GainControlImpl into the GainControlConfigProxy, as it becomes the
   sole AGC object with functionality exposed to the client.

Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
2019-03-27 15:19:41 +00:00
Per Åhgren
03257b049e Add flag for explicitly specifying that the legacy AEC2 should be used
This CL adds a temporary flag for specifying that the legacy AEC2 should
be used.

Bug: webrtc:10366
Change-Id: Ie3edaa1560cdc1282b62242beb67aa6fee7f2841
Reviewed-on: https://webrtc-review.googlesource.com/c/124980
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26891}
2019-02-28 10:56:27 +00:00
Mirko Bonadei
d970807e0c Remove rtc_base/scoped_ref_ptr.h.
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.

More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.

Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
Sam Zackrisson
235131303b Add noise suppression settings to AudioProcessing::Config
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}
2019-01-14 16:17:19 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Sam Zackrisson
4db667be74 Add private voice detection instance to replace public voice detector
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct.

The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable().

Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible.

Bug: webrtc:9947
Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d
Reviewed-on: https://webrtc-review.googlesource.com/c/115243
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26101}
2018-12-27 12:00:06 +00:00
Sam Zackrisson
b24c00f02d Add AudioProcessingCaptureStats and a level estimator replacement
This adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.

Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}
2018-11-26 15:52:14 +00:00
Alessio Bazzica
8da7b350cf AGC2 adaptive digital false by default
Avoid that the client code relies on the adaptive digital mode being
enabled by default (error prone).

Bug: webrtc:7494
Change-Id: I765fecf535cf31a2163e10595a42520473c233b6
Reviewed-on: https://webrtc-review.googlesource.com/c/111586
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25728}
2018-11-21 14:20:15 +00:00
Alessio Bazzica
dd886082c5 AGC2 flags: remove deprecated fields.
Downstream projects adapted, clean up.

Bug: webrtc:7494
Change-Id: I019b8dd79c6bc55c900fb5595d5e2ee8aa0a2400
Reviewed-on: https://webrtc-review.googlesource.com/c/110865
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25656}
2018-11-15 13:47:24 +00:00
Alessio Bazzica
1e2542f593 AGC2: adding level estimation option (RMS or peak-based).
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}
2018-11-13 14:32:13 +00:00
Sam Zackrisson
63ada787b5 Remove outdated TODO
Bug: webrtc:9535
Change-Id: I8f7a719eb9f32a91f45620453568e5f7d2264de8
Reviewed-on: https://webrtc-review.googlesource.com/c/109461
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25511}
2018-11-06 08:39:16 +00:00
Alex Loiko
5e784616e0 Make the extra seturation margin configurable.
The extra saturation margin is a setting for the SaturationProtector
in GainController2. The higher it is, the less gain GC2 will apply. In
this CL we pipe the setting up to audio_processing.h. Now the setting
can be set at a high level.

Also in this CL add a few (missing, they should have been there
already) tests for the GC2 and GC2 with saturation margin.

Bug: webrtc:7494
Change-Id: I1b61f1662e6c6a8817fd5b0e845339694bf8d50d
Reviewed-on: https://webrtc-review.googlesource.com/c/109001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25470}
2018-11-01 15:12:11 +00:00