Commit graph

127 commits

Author SHA1 Message Date
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
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
Gustaf Ullberg
3c918b1af8 Fix bypass of unnecessary resampling
This change fixes an issue with bypass of unnecessary resampling
when using ProcessStream(AudioFrame*).

Bug: b/130016532
Change-Id: I887f05d55aaa47f21164ba237cf83d0be33a1fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156540
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29446}
2019-10-11 11:57:36 +00:00
Gustaf Ullberg
8675eeec26 Bypass unnecessary resampling.
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}
2019-10-10 08:38:41 +00:00
Gustaf Ullberg
422b9e0982 Run fullband processing at output rate on ARM
The audio processing in the band-split domain on ARM platforms
operate at a sampling frequency of 32 kHz. This CL upsamples
the signal to fullband before the "fullband processing"
if an output rate of 48 kHz is chosen.

Change-Id: I268acd33aff1fcfa4f75ba8c0fb3e16abb9f74e8
Bug: b/130016532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155640
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29415}
2019-10-09 12:09:06 +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
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +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
ce202a0f98 Reland "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
This is a reland of a66395e72f

Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
> 
> This is a reland of f3a197e553
> 
> Original change's description:
> > 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}
> 
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}

Bug: webrtc:10913
Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29042}
2019-09-03 06:12:32 +00:00
Per Åhgren
d112c75801 Revert "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
This reverts commit a66395e72f.

Reason for revert: Breaking downstream tests

Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
> 
> This is a reland of f3a197e553
> 
> Original change's description:
> > 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}
> 
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}

TBR=saza@webrtc.org,peah@webrtc.org

Change-Id: I0e9fd154da5910d73b7a4c82e4e588f3220fd39d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151126
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29038}
2019-09-02 13:57:07 +00:00
Per Åhgren
a66395e72f Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
This is a reland of f3a197e553

Original change's description:
> 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}

Bug: webrtc:10913
Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29034}
2019-09-02 12:08:27 +00:00
Ilya Nikolaevskiy
e3e30ae5c5 Revert "Add core multi-channel pipeline in AEC3"
This reverts commit f3a197e553.

Reason for revert: Speculative revert, as this may'be broken some build bots

Original change's description:
> 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}

TBR=saza@webrtc.org,peah@webrtc.org

Change-Id: I877d2993b9ccf024bd1d57bca1513c3e24d0bed3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29022}
2019-08-30 10:19:29 +00:00
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
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
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
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
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
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
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
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
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
Per Åhgren
f204fafdb4 Only create AEC2 when needed
This CL ensures that the AEC2 is only created when needed.
The changes in the CL are bitexact when running AEC2 via
audioproc_f

Bug: webrtc:8671
Change-Id: I5f6d33e45a7031c69ac53098781635c415668e49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129740
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27772}
2019-04-25 14:01:12 +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
200feba1c0 Make AEC3 the default desktop AEC option in WebRTC
Bug: webrtc:10366
Change-Id: I854ed62df1da489fdab43e9157dff79b7287cacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125081
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26983}
2019-03-06 08:43:48 +00:00
Sam Zackrisson
421c859351 Remove crit_render_ lock from webrtc::GainControlImpl
The lock is unnecessary and potentially unsafe:
1) All gain_control accesses in AudioProcessingImpl happen - and are intended to happen - while holding the crit_capture_ lock, and all external API calls take the same lock once inside GainControlImpl.
2) If ProcessCaptureStreamLocked (locked by crit_capture) calls a gain_control function that takes crit_render, the mandated locking order (render before capture) is violated and we might get a deadlock with the render thread.

Bug: b/123456404
Change-Id: Id7a888827e347e5e1d50e2f87d90e8b68f52b7b8
Reviewed-on: https://webrtc-review.googlesource.com/c/122087
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26637}
2019-02-11 14:14:40 +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
Sam Zackrisson
c22f551842 Remove locks from AECM and move it into private_submodules_
This drops the locks and annotations in EchoControlMobileImpl,
now that the interface is no longer externally accessible.

Additionally, SetEchoPath and GetEchoPath (with surrounding code) is
removed. They are unused.

Bug: webrtc:9929
Change-Id: Ibc6751754614ed39836f6ee6835d7b53dedd828c
Reviewed-on: https://webrtc-review.googlesource.com/c/109025
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25504}
2018-11-05 16:25:09 +00:00
Sam Zackrisson
281276301c Remove deprecated AudioProcessing::GetStatistics function
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}
2018-11-01 11:21:15 +00:00
Sam Zackrisson
7f4dfa4106 Remove locks from AEC2 and move it into private_submodules_
This drops the locks and annotations in EchoCancellationImpl,
now that the interface is no longer externally accessible.


Bug: webrtc:9929
Change-Id: I401256f523340cbabce23a5914ab28ce44179935
Reviewed-on: https://webrtc-review.googlesource.com/c/108602
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25460}
2018-11-01 08:40:16 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Sam Zackrisson
b0ab2ce256 Reland "Remove the HighPassFilter interface"
Downstream Chromium dependencies fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/1286449

This is a reland of e2405c1a82

Original change's description:
> Remove the HighPassFilter interface
>
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
>
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

Bug: webrtc:9535
Change-Id: I0017193ad3ca1762e186f3ad79f29d33ef468202
Reviewed-on: https://webrtc-review.googlesource.com/c/106681
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25300}
2018-10-23 07:44:09 +00:00
Niklas Enbom
d895f42bfb Revert "Remove the HighPassFilter interface"
This reverts commit e2405c1a82.

Reason for revert: Breaks Chrome compile: https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8932502586827763408/+/steps/compile__with_patch_/0/stdout 
Original change's description:
> Remove the HighPassFilter interface
> 
> The functionality remains unaffected.
> Filter toggling is still available via webrtc::AudioProcessing::Config.
> Example:
> webrtc::AudioProcessing::Config config = apm.GetConfig();
> // Read settings
> if (config.high_pass_filter.enabled) { ... }
> // Apply setting
> config.high_pass_filter.enabled = true;
> apm.ApplyConfig();
> 
> Bug: webrtc:9535
> Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
> Reviewed-on: https://webrtc-review.googlesource.com/c/102541
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25198}

TBR=solenberg@webrtc.org,saza@webrtc.org,peah@webrtc.org

Change-Id: Ieb34d5c573c4ab22eefbb54aeaa2f72844740b89
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/106421
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25215}
2018-10-16 15:51:45 +00:00
Sam Zackrisson
e2405c1a82 Remove the HighPassFilter interface
The functionality remains unaffected.
Filter toggling is still available via webrtc::AudioProcessing::Config.
Example:
webrtc::AudioProcessing::Config config = apm.GetConfig();
// Read settings
if (config.high_pass_filter.enabled) { ... }
// Apply setting
config.high_pass_filter.enabled = true;
apm.ApplyConfig();

Bug: webrtc:9535
Change-Id: Ib4c4b04078bbb490ebdab9721b8c7811d73777a8
Reviewed-on: https://webrtc-review.googlesource.com/c/102541
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25198}
2018-10-16 09:27:44 +00:00
Yves Gerey
499bc6c5d0 Fix race conditions for ReofferDoesNotCallOnTrack test.
This CL extend critical sections to incorporate:
 * private_submodules_->echo_controller
 * config_

As a side benefit, it prevents weird interleaving where configuration
could have been changed in the middle of GetStatistics methods.

Bug: webrtc:9841
Change-Id: I0de5e756a684c2ff1be4effccf8c0f3d3175e3b9
Reviewed-on: https://webrtc-review.googlesource.com/c/105142
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25121}
2018-10-11 16:12:12 +00:00
saza
be490b2abe Delete deprecated AEC interfaces
They've been officially deprecated since September 4, 2018.
PSA: https://groups.google.com/forum/#!topic/discuss-webrtc/r_9n-PRUIX4

Bug: webrtc:9535
Change-Id: I294e22ae874b1edd81a0a0347755d82c5ebc61e0
Reviewed-on: https://webrtc-review.googlesource.com/c/103444
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24971}
2018-10-04 09:20:10 +00:00
Sam Zackrisson
8c147b68e6 Reland "Remove APM-internal usage of EchoControlMobile"
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}
2018-10-03 07:45:33 +00:00