Commit graph

712 commits

Author SHA1 Message Date
Per Åhgren
f3a197e553 Add core multi-channel pipeline in AEC3
This CL adds basic the basic pipeline to support multi-channel
processing in AEC3.

Apart from that, it removes the 8 kHz processing support in several
places of the AEC3 code.

Bug: webrtc:10913
Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29017}
2019-08-30 08:07:27 +00:00
Per Åhgren
3e8bf282c4 Increase the maximum supported sample rate to 384000 Hz and add tests
This CL increases the maximum supported sample rate so that all rates
up to 384000 Hz are handled.

The CL also adds tests that verifies that APM works as intended for
different combinations of number of channels and sample rates.

Bug: webrtc:10882
Change-Id: I98738e33ac21413ae00fec10bb43b8796ae2078c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150532
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29014}
2019-08-29 22:14:25 +00:00
Oleksandr Iakovenko
d191533717 Fix wrong-import-order pylint errors in quality_assessment.signal_processing module.
Bug: webrtc:10924
Change-Id: I9aeb062f1e4b4e061192b42b9a1f8b061fbbb8c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150646
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28989}
2019-08-28 14:48:28 +00:00
Sam Zackrisson
6558fa5d64 Reintroduce command line controlled reference data updating for ApmTest.Process
Replaces a hardcoded bool in a test with command line flag.

The current hardcoding of the bool is a little bit hacky. And the
tests will pass automatically, so it is possible to accidentally
commit the flipped bool in a CL, like here:
https://webrtc-review.googlesource.com/c/src/+/150221

I am fairly sure this resolves the vague issue referred to in the attached bug.
The bug is introduced with a TODO here:
https://webrtc-codereview.appspot.com/1728005/diff/4001/webrtc/modules/audio_processing/test/unit_test.cc
Another TODO was added later that refers to the first TODO:
https://webrtc-codereview.appspot.com/6879004/diff/150001/webrtc/modules/audio_processing/test/audio_processing_unittest.cc

Bug: webrtc:1981
Change-Id: I066f41add602c791a5f2ba18829c4306da7dac15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150334
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28955}
2019-08-26 11:12:55 +00:00
Per Åhgren
2ca0b3689f Correct the handling of sample rates that don't scale well into even 10 ms chunks
This CL corrects the way the audio processing module handles sample rates that
don't allow partitioning the data into evenly sized 10 ms chunks, examples
being 22050 Hz and 11025 Hz.

Bug: webrtc:10882
Change-Id: I35d738f8a0e1debc443fe5d473c0d666a7ba8d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150526
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28953}
2019-08-26 09:54:48 +00:00
Per Åhgren
0aefbf0ec4 Use the AEC3 high-pass filter for the whole APM
This CL removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.

Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
2019-08-23 20:04:10 +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
Steve Anton
f254e9e9e5 Revert "Simplification and refactoring of the AudioBuffer code"
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}
2019-08-21 18:00:59 +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
Per Åhgren
81c0cf287c 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}
2019-08-21 13:40:59 +00:00
Mirko Bonadei
2dac4e4e35 Remove rtc_use_lto GN arg.
This CL is a no-op since rtc_use_lto is always false and in general
such change should probably be implemented in
//build/config/compiler/BUILD.gn.

Bug: chromium:408997
Change-Id: Id37d3181e66e699f8cd535aee1af7609352a7259
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149833
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28919}
2019-08-20 14:00:49 +00:00
Per Åhgren
928146f546 Removing all external access to the integer sample data in AudioBuffer
This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.

The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.

Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
2019-08-20 08:36:47 +00:00
Per Åhgren
62c174c5a1 Reland of Correct conversion between float and fixed formats
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.

In contrast to the initial CL, the DCHECKS on the incoming sample
range was changed to limiting.

Bug: webrtc:6594
Change-Id: I8218dfd5c45388ad5aac61be453d2f28725a2475
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#28867}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149169
Cr-Commit-Position: refs/heads/master@{#28897}
2019-08-19 12:15:07 +00:00
Gustaf Ullberg
cd277b84da AEC3: Fix computation of audio buffer delay
This change fixes a bug where the initial delay could be set incorrectly.

Bug: webrtc:10896
Change-Id: I66b2234b69c46639488f4561e973384001230861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149820
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28894}
2019-08-19 11:05:21 +00:00
Henrik Andreassson
533c225c93 Revert "Correct conversion between float and fixed formats"
This reverts commit 67e43c8b95.

Reason for revert: speculative revert since we see failing bots on Android after this change

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/4124

Original change's description:
> Correct conversion between float and fixed formats
> 
> This CL changes the way that values are converted
> between fixed and floating point to
> -Avoid the former asymmetric conversion causing
> nonlinear distortions.
> -Reduce the complexity.
> 
> Bug: webrtc:6594
> Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28867}

TBR=henrik.lundin@webrtc.org,gustaf@webrtc.org,peah@webrtc.org

Change-Id: Id828a09de7075e48556fe2d0beba7a0c6ec227f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6594
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149165
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28872}
2019-08-16 10:40:11 +00:00
Per Åhgren
67e43c8b95 Correct conversion between float and fixed formats
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.

Bug: webrtc:6594
Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28867}
2019-08-15 13:51:39 +00:00
Per Åhgren
a1351271e6 Remove all AudioBuffer code that is not related to storing audio data
This CL moves/removes all code from the AudioBuffer that:
-Is not directly handling audio data (e.g., keytaps, VAD descisions).
-Is caching aggregated versions of the rest of the audio data.
-Is not used (or only used in testing)

Bug: webrtc:10882
Change-Id: I737deb3f692748eff30f46ad806b2c6f6292802c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149072
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28866}
2019-08-15 13:49:29 +00:00
Sonia-Florina Horchidan
b75d14c802 audioproc_f: input AEC dump as string, output audio to vector
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}
2019-08-12 09:17:36 +00:00
Gustaf Ullberg
940c2b5005 AEC3: Reduce level of log messages
This change reduces the level of several non-critical log messages in
order to reduce log spamming.

Bug: webrtc:8671
Change-Id: I6faae7a2ae4eeafd18c2770208485a75ad946e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148528
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28809}
2019-08-08 14:08:05 +00:00
Per Åhgren
eac47f7fae Removing unused fallback variant for the reverb computation
This CL removes a long unused fallback behavior for the reverb
computation.

Bug: webrtc:8671
Change-Id: I4b57795a9bb33769237858f40392881ee235653e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148520
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28802}
2019-08-08 08:00:38 +00:00
Per Åhgren
45231be79c AEC3: Removing unused code in the echo subtractor
Bug: webrtc:8671
Change-Id: I77e9c55fe2e1030e5b74c02d4bc9222de422f6f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137045
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28760}
2019-08-05 14:53:59 +00:00
Per Åhgren
cdbaeeb737 Aec3:Remove unused legacy code
Bug: webrtc:8671
Change-Id: I54f14cc304dbe1639fbb356dbdf9beeb1781ede4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137801
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28759}
2019-08-05 14:13:54 +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
Niels Möller
74cebb3c6d Mark the audio_generator_factory target as testonly
Bug: None
Change-Id: Iad9b740835e373b254c9702f8178b4bb41eb36c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147266
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28706}
2019-07-30 13:50:54 +00:00
Mirko Bonadei
2ab97f6f8e Migrate WebRTC test infra to ABSL_FLAG.
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}
2019-07-19 06:54:04 +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
Oleh Prypin
eec86cdd96 Fix platform-specific header dependencies to be more precise
Make the GN conditionals match what happens in sources, or the other way around. Include headers only when they're used.

Bug: None
Change-Id: Ib8e3346e3c24eaa7e61ac4776dcd66efe2cc5c65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28500}
2019-07-08 11:21:30 +00:00
Mirko Bonadei
9eee121a8f Switch py_quality_assessment to ABSL_FLAG.
Bug: webrtc:10616
Change-Id: I051d5706576d5684d82e3e42fb1b40ea755864d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144054
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28464}
2019-07-03 12:46:18 +00:00
Mirko Bonadei
bc70b6164e Switch rnn_vad_tool to ABSL_FLAG.
Tested:
$ ./out/Debug/rnn_vad_tool --i ./data/voice_engine/audio_tiny8.wav \
  --o /tmp/o.prob --f /tmp/o.feat
(rnn_vad_tool.cc:47): Input sample rate: 8000
(rnn_vad_tool.cc:105): VAD probabilities written to /tmp/o.prob
(rnn_vad_tool.cc:108): features written to /tmp/o.feat

Bug: webrtc:10616
Change-Id: Ied33d9425bc1621d084bb04d9acf12ea9602a88b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144048
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28445}
2019-07-02 12:27:15 +00:00
Karl Wiberg
225842ced8 Initialize signal processing function pointers statically
The last run-time logic for selecting function pointers was removed in
May 2016, here: https://codereview.webrtc.org/1955413003

It would be even better if we could eliminate the function pointers
entirely and just have different implementations that we select at
compile time; I've left a TODO asking for this.

Bug: webrtc:9553
Change-Id: Ica71d71e19759da00967168f6479b7eb8b46c590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144053
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28414}
2019-06-28 14:20:03 +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
Niels Möller
e8e4dc4c8b Change StartAecDump methods to work with FILE* and FileWrapper
Bug: webrtc:6463
Change-Id: Id275975decb9b2876021ced19ee9f279b07bea53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140283
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28233}
2019-06-11 13:43:36 +00:00
Niels Möller
646e096e03 Allow null FILE* to AecDumpFactory::Create
Also add Create method with a FileWrapper argument, extracted from cl
https://webrtc-review.googlesource.com/c/src/+/140283

Bug: webrtc:6463
Change-Id: If86f17d5fda964df0b5874bd5a15fcaed899dd4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140946
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28211}
2019-06-10 14:09:19 +00:00
Mirko Bonadei
71061bcca8 Replace calls to deprecated googletest APIs.
SetUpTestCase/TearDownTestCase -> SetUpTestSuite/TearDownTestSuite.

TBR=kwiberg@webrtc.org

Bug: None
Change-Id: I6d873c62d6b5c9d7100624d00e1c4894d686a9f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140041
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28189}
2019-06-07 06:41:20 +00:00
Oskar Sundbom
4b27648d8b Avoid the render lock in AudioProcessingImpl::ProcessStream
It seems unnecessary to lock it if not actually reinitializing.

Bug: webrtc:10205
Change-Id: Ib3292e1d640a92a7df77400aebe9583cf877f824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/115460
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28060}
2019-05-24 13:24:27 +00:00
Alessio Bazzica
b9979a533c AGC2 RNN VAD: remove unused dep (KissFFT)
NOTRY=True

Bug: webrtc:9577,webrtc:10480,webrtc:9139
Change-Id: I9fdf8c3bfd91d11fe01860546bcb83a78f5443fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137434
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27989}
2019-05-20 14:43:33 +00:00
Per Åhgren
67c76b214d AEC3: Minor code corrections
Bug: webrtc:8671
Change-Id: I096053087e7ef0f3375f9c20b55558c1cec670cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136806
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27963}
2019-05-16 13:53:27 +00:00
Per Åhgren
46ac470d92 AEC3: Remove unused code
Bug: webrtc:8671
Change-Id: Ia54cc2eb7f7235cd348a1af216beb3981fac9813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136580
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27928}
2019-05-13 13:29:45 +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
Oleh Prypin
199295882d Qualify cmath function calls
Use the C++-style stdlib headers, add `std::` prefix, in order to avoid implicit casts to double.

Bug: None
Change-Id: I78d9caaee715be341d2480c6d5e769068966d577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133625
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27905}
2019-05-10 09:00:54 +00:00
Fredrik Hernqvist
bf47f340ee Add comments to clarify argument meanings in APM impl test
Bug: webrtc:10608
Change-Id: Iac1111b739458a1b0ce1cac5e59de06905c085d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135574
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27891}
2019-05-09 09:22:17 +00:00
Per Åhgren
d547d862d5 Remove the enable flag from AEC2 and AECM
This CL removes the redundant enable flags from AEC2 and AECM

Bug: webrtc:5298
Change-Id: Icc575abf1c368dda02ca77f057d166f1c921f662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135100
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27848}
2019-05-03 14:20:32 +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
Niels Möller
4731f0062e Delete deprecated PlatformThread looping
Bug: webrtc:10594, webrtc:7187
Change-Id: Icba3a5cf6dbe817ead427c27645b3ad7bc8819be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134642
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27833}
2019-05-03 08:35:42 +00:00
Alessio Bazzica
c0c7d36e80 RNN VAD: clean-up unit tests
- add test that checks that the computed VAD probability is within
  tolerance *1
- speed-up some tests by reducing the input length and skipping frames
- remove unused code in test_utils
- fix some comments

*1: RnnVadTest::RnnBitExactness is replaced by
    RnnVadTest::RnnVadProbabilityWithinTolerance

Bug: webrtc:10480
Change-Id: I19332d06eacffbbe671bf7749ff4c92798bdc55c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133910
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27803}
2019-04-29 12:55:02 +00:00
Per Åhgren
b6e24d7f35 Only create AECm when needed
This CL ensures that the AECm is only created when needed.
The changes in the CL are bitexact when running AECm via
audioproc_f

The CL also corrects an issue where there is a risk for
AEC2 to not be correctly setup when the sample rate
changes inbetween activations.

Bug: webrtc:8671
Change-Id: Id3b33e20969b1543e28c885d47495246cfbe549d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134216
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27800}
2019-04-29 10:51:51 +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