214 Commits

Author SHA1 Message Date
Niels Möller
cccd55094d Delete unneeded dependencies on deprecated build target webrtc_common
Bug: webrtc:7660
Change-Id: Iad32aad8432fa2c6b3018d511b51943f869fbd11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182420
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31986}
2020-08-25 07:33:12 +00:00
Danil Chapovalov
70b2cf8b36 Delete deprecated version of EncodedImageCallback::OnEncodedImage
Bug: webrtc:6471
Change-Id: I173cd3b3b9f4badaf7c17574adf1d09a926a9b9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31982}
2020-08-24 11:00:19 +00:00
Niels Möller
afadfb24a5 Delete CodecInfo::is_hardware_accelerated
Followup to https://webrtc-review.googlesource.com/c/src/+/179520

Bug: None
Change-Id: I083573ec977f80437f59549358069df6876f3d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31961}
2020-08-19 09:42:30 +00:00
Niels Möller
5b69aa6613 Move definition of SpatialLayer to api/video_codecs/spatial_layer.h
Bug: webrtc:7660
Change-Id: I54009ebc5f65b6875a8c079ab5264e0c5ce9f654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31942}
2020-08-17 09:45:19 +00:00
Niels Möller
b08177353b Delete VideoCodec::plType member
Followup to https://webrtc-review.googlesource.com/c/src/+/181201.

Bug: None
Change-Id: Ia7e2a924d2a192472f4a9cd2572e3443a313e41a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181370
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31929}
2020-08-13 14:18:16 +00:00
Niels Möller
5401bad701 Prepare for deleting VideoCodec::plType
Deletes all webrtc usage of this member. Next step is to delete
any downstream references, and when that's done, the member can be
deleted.

Bug: None
Change-Id: I3f3a94a063dccf56468a1069653efd3809875b01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181201
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31911}
2020-08-11 14:20:59 +00:00
Danil Chapovalov
0490c37126 Reland "Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader"
This is a reland of fdd41ecf6b05b02a1ee12583e537f1cdb5922919

Original change's description:
> Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader
> 
> RTPFragmentationHeader is no longer used by any webrtc code.
> 
> Bug: webrtc:6471
> Change-Id: I1362d58465c550f5c8e0a5b292dc0abcd6251431
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180120
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31833}

Bug: webrtc:6471
Change-Id: I7f551110a68704bfe12897772083c3616acd173c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180881
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31848}
2020-08-04 12:17:06 +00:00
Florent Castelli
d3511010d9 Reland "Only enable conference mode simulcast allocations with flag enabled"
This is a reland of 32ca95145c4636374266f5b5d4d1ac43658bc758

Fix includes not enabling the screenshare conference behavior on non
screenshare sources even if the flag is enabled.

Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}

Bug: webrtc:11310
Bug: chromium:1093819
Change-Id: Ic933f93a5c4bad20583354fe821f8a1170e911cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180802
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31847}
2020-08-04 10:30:08 +00:00
Niels Möller
2b781bf908 Deprecate write-only member CodecInfo::is_hardware_accelerated
This member of the CodecInfo struct was set in several places, but not
used for anything. To aid deletion, this cl defines a default implementation
of VideoEncoderFactory::QueryVideoEncoder.

The next step is to delete almost all downstream implementations of that method,
since the only classes that have to implement it are the few factories that
produce "internal source" encoders, e.g., for Chromium remoting.

Bug: None
Change-Id: I1f0dbf0d302933004ebdc779460cb2cb3a894e02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31844}
2020-08-04 07:56:49 +00:00
Marina Ciocea
65e3044b40 Revert "Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader"
This reverts commit fdd41ecf6b05b02a1ee12583e537f1cdb5922919.

Reason for revert: Breaks WebRTC roll into Chrome: 
https://crrev.com/c/2335265

Sample failure:
https://ci.chromium.org/p/chromium/builders/try/linux_chromium_compile_dbg_ng/688153?


Original change's description:
> Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader
> 
> RTPFragmentationHeader is no longer used by any webrtc code.
> 
> Bug: webrtc:6471
> Change-Id: I1362d58465c550f5c8e0a5b292dc0abcd6251431
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180120
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31833}

TBR=danilchap@webrtc.org,brandtr@webrtc.org,nisse@webrtc.org,srte@webrtc.org

Change-Id: I6bd021637b0c140f6625612caebdb5829decdfb0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180821
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31839}
2020-08-03 18:31:07 +00:00
Danil Chapovalov
fdd41ecf6b Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader
RTPFragmentationHeader is no longer used by any webrtc code.

Bug: webrtc:6471
Change-Id: I1362d58465c550f5c8e0a5b292dc0abcd6251431
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180120
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31833}
2020-08-03 13:58:17 +00:00
Florent Castelli
834dc9cfa1 Revert "Only enable conference mode simulcast allocations with flag enabled"
This reverts commit 32ca95145c4636374266f5b5d4d1ac43658bc758.

Reason for revert: Internal test failure

Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
> 
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
> 
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
> 
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}

TBR=ilnik@webrtc.org,hta@webrtc.org,orphis@webrtc.org

Change-Id: I5ccb6e87594f491ba09fe6b837ee24d63db878ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11310
Bug: chromium:1093819
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180801
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31829}
2020-08-03 10:31:21 +00:00
Florent Castelli
32ca95145c Only enable conference mode simulcast allocations with flag enabled
Non-conference mode simulcast screenshares were mistakenly using the
conference mode semantics in the simulcast rate allocator, which broke
spec compliant usage in some situation.

This behavior should only be used when explicitly using the SDP entry
"a=x-google-flag:conference" in both offer and answer.

Bug: webrtc:11310, chromium:1093819
Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31828}
2020-08-03 10:09:46 +00:00
Philip Eliasson
49c293f03d Revert "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
This reverts commit 4ba1044bae750ab8ee47b359c21f672386b7c3cd.

Reason for revert: Downstream projects require some updates.

Original change's description:
> Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id.
> 
> Bug: webrtc:9106
> Change-Id: I7fa84095732c33d136a9354ae4f09266cffcf877
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180020
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31793}

TBR=henrika@webrtc.org,magjed@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org

Change-Id: I8c980266334cc9871b9076713da3c4df8f73f8ce
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9106
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180344
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31794}
2020-07-27 13:55:00 +00:00
philipel
4ba1044bae Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id.
Bug: webrtc:9106
Change-Id: I7fa84095732c33d136a9354ae4f09266cffcf877
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180020
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31793}
2020-07-27 13:26:52 +00:00
Niels Möller
3592839896 Add default values for VideoEncoderFactory::CodecInfo
So that applications can construct a default struct without naming the
members.

Bug: None
Change-Id: Idd9028bee9016670e776f17a62077eb9c34d6f2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179485
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31742}
2020-07-16 11:12:35 +00:00
Mirko Bonadei
24762f207f Fix missing dependencies.
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.

Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
2020-06-23 15:46:34 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
Jerome Jiang
7f7fb830ba Reland "Add av1 test running real video clips."
This reverts commit 6958d2c6f0ce5267bdc4120d88680a4be9ed5e59.

Disable the test on iOS.

Bug: None
Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31437}
2020-06-04 06:32:46 +00:00
Ying Wang
6958d2c6f0 Revert "Add av1 test running real video clips."
This reverts commit 3a2be87b80759a659901776024609de7a19fbaca.

Reason for revert: break internal test

Original change's description:
> Add av1 test running real video clips.
> 
> Bug: None
> Change-Id: I93bb8b3bf15d607d061aa74ad9e34609ffb2ef0a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175821
> Commit-Queue: Jerome Jiang <jianj@google.com>
> Commit-Queue: Stefan Holmer <holmer@google.com>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31401}

TBR=danilchap@webrtc.org,jianj@google.com,stefan@webrtc.org,holmer@google.com,marpan@webrtc.org

Change-Id: I2689ab4f7f26af6e26a4a188a2aa0b4f90a1a92f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176374
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31405}
2020-06-02 10:40:38 +00:00
Jerome Jiang
3a2be87b80 Add av1 test running real video clips.
Bug: None
Change-Id: I93bb8b3bf15d607d061aa74ad9e34609ffb2ef0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175821
Commit-Queue: Jerome Jiang <jianj@google.com>
Commit-Queue: Stefan Holmer <holmer@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31401}
2020-06-02 07:36:20 +00:00
Ilya Nikolaevskiy
35fc1537af Fallback to software decoders on consequtive decode errors on key-frames
Bug: webrtc:11575
Change-Id: I09be17ab5155e9f610c8f7c451ca52d7d65e24d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175222
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31295}
2020-05-18 09:57:04 +00:00
Mirko Bonadei
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2).
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
2020-04-02 14:38:30 +00:00
Johannes Kron
570330361a Add fallback histograms for VideoDecoderSoftwareFallbackWrapper
Track the number of samples that are decoded until a fallback to
software decoder happens.

Bug: chromium:1061376
Change-Id: Ida3ae94034ec83a6d28001cb7be343b8b99b99c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170468
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30814}
2020-03-17 14:55:24 +00:00
Henrik Boström
b0f2e0ced4 [Overuse] Make VideoStreamAdapter responsible for executing adaptation.
This CL moves GetAdaptUpTarget(), GetAdaptDownTarget() and
ApplyAdaptationTarget() - and related code - to the VideoStreamAdapter.

This includes pieces related to calculating how to adapt, including:
- DegradationPreference
- BalancedDegradationPreference
- AdaptationRequest and last_adaptation_request_
- CanAdaptUpResolution()

The VideoStreamAdapter's interface has changed: VideoSourceRestrictor
methods are now hidden in favor of methods exposing AdaptationTarget.

This CL also does some misc moves:
- GetEncoderBitrateLimits is moved and renamed to
  VideoEncoder::EncoderInfo::GetEncoderBitrateLimitsForResolution.
- EncoderSettings moved to a separate file.

// For api/video_codecs/video_encoder.[cc/h] changes, which is the
// moving of a function.
TBR=sprang@webrtc.org

Bug: webrtc:11393
Change-Id: Ie6bd8ef644ce927d7eca6ab90a0a7bcace682f3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169842
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30708}
2020-03-06 13:35:20 +00:00
Mirta Dvornicic
4f34d78c85 Report available instead of encoding bitrate to VideoEncoderSelector.
The encoding bitrate might be limited depending on the current encoder.

Bug: webrtc:11341
Change-Id: I734fce12734b1e703e7948847cdb1365c08a137b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169123
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30619}
2020-02-26 15:56:36 +00:00
Danil Chapovalov
cad3e0e2fa Replace DataSize and DataRate factories with newer versions
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
2020-02-18 16:09:50 +00:00
Evan Shrubsole
546a9e4350 Scale native frames when doing a SW codec fallback
If the incoming frame is a native frame but the native encoder fails,
we should ensure the fallback encoder can handle the native frame. If
not then the native frame should be scaled and converted.

Bug: webrtc:11346
Change-Id: I692350dc69b5ce2db7ba5ee98d28f94cb12054cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168345
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30504}
2020-02-12 08:55:51 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 12:21:17 +00:00
philipel
9b05803e19 Implement injectable EncoderSelectorInterface and wire it up in the VideoStreamEncoder.
The EncoderSelectorInterface is meant to replace the "WebRTC-NetworkCondition-EncoderSwitch" field trial, so the field trial will be ignored if an EncoderSelectorInterface object has been injected.

Bug: webrtc:11341
Change-Id: I5371fac9c9ad8e38223a81dd1e7bfefb2bb458cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168193
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30490}
2020-02-10 12:12:47 +00:00
Erik Språng
261f792f83 Allow software fallback on lowest simulcast stream for temporal support
Bug: webrtc:11324
Change-Id: Ie505be0cda74c0444065d86c3727671c62bd4842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167527
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30437}
2020-01-31 16:44:47 +00:00
Rasmus Brandt
43bfe0b8a6 Enforce VideoEncoderConfig.num_temporal_layers >= 1.
This change clarifies the semantics of this field:
  unset: Depends on context.
  == 0: Invalid.
  == 1: No temporal layering.
  >= 2: Temporal layering.

We should try to remove the wrapping optional later.

Bug: webrtc:11297
Change-Id: Id765f2dc1d31a4ba3cd424978ac6054cd60152ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166528
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30336}
2020-01-21 13:38:08 +00:00
Henrik Boström
4bab2fcf6b [Overuse] Setting encoder configurations through the interface.
This squashes together several input signals that were spread out
through several calls into a single method and calling place:
SetEncoderSettings(), invoked from ReconfigureEncoder(). This is added
to the abstract interface.

This makes the following methods obsolete which are removed:
- SetEncoder(): The VideoEncoder was only used for GetEncoderInfo();
  the VideoEncoder::EncoderInfo is now part of the EncoderSettings.
- SetEncoderConfig(): The VideoEncoderConfig is part of
  EncoderSettings. The config is used for its codec_type and
  content_type enums.
- SetCodecMaxFrameRate(): The max frame rate was the same as
  VideoCodec::maxFramerate. VideoCodec is now part of EncoderSettings.

There may be some overlap in information between EncoderConfig and
VideoCodec, but that is outside the scope of this CL, which only makes
sure to bundle encoder settings-like information into one input signal.

Bug: webrtc:11222
Change-Id: I67c49c49c0a859cb7d5051939a461593c695a789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166602
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30332}
2020-01-21 11:48:11 +00:00
Danil Chapovalov
64f1f3f04e Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED
Bug: None
Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 15:20:35 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
Rasmus Brandt
5cad55b240 Signal requested resolution alignment requirements from sinks to sources.
Bug: webrtc:11218
Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30113}
2019-12-19 10:39:04 +00:00
Artem Titov
8525a8028a Add ability to resize buffers pool in decoder and use it in IVF generator
Bug: webrtc:10138
Change-Id: I452f08f1d9af57de789bd947a1fcb95536845f80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162183
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30098}
2019-12-16 14:51:16 +00:00
Hirokazu Honda
1154915024 video_encoder: Add RTC_EXPORT ScalingSettings
ScalingSettings has some public constructors. These should be
able to be called from exteranl code. However, a linker fails
on windows because ScalingSettings doesn't have RTC_EXPORT.

Bug: chromium:1031965
Test: build crrev.com/c/1949841
Change-Id: Iddaea77f87c52edbe8f77551322d7aa198bc0aeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30063}
2019-12-11 15:14:56 +00:00
Florent Castelli
907dc806c7 Reland "Add support for RtpEncodingParameters::max_framerate"
Perf test failure was fixed separately.

TBR=steveanton@webrtc.org,sprang@webrtc.org,asapersson@webrtc.org

Original change's description:
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

Bug: webrtc:11117
Change-Id: I9c1daf7887c2024c6669dc79bff89d737417458c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161445
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30030}
2019-12-06 15:11:54 +00:00
Erik Språng
7968530418 Removes caching SimulcastEncoderAdapter::GetEncoderInfo()
There are edge cases where the caching of encoder info will cause
issues. For instance if a sub-encoder fails en Encode call and falls
back to some other implementation, or if the fps targets shift due to
SetRates() triggering new layers to be enabled.

This CL forces a complete rebuild on every call to GetEncoderInfo().

It also adds new logging of when the info changes, as debugging issues
can be very time consuming if we can't tell that happened.

Bug: webrtc:11000
Change-Id: I7ec7962a589ccba0e188e60a11f851c9de874fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29938}
2019-11-27 17:06:54 +00:00
Danil Chapovalov
b529b7aeba Add string<->VideoCodecType conversion for all codec types.
Use that conversion instead of duplicating it in call/

Bug: webrtc:11042
Change-Id: I035b161d429ec339dd2ad9e9ed3ede5045fb6199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160881
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29936}
2019-11-27 14:15:07 +00:00
Florent Castelli
5cef9c3581 Revert "Add support for RtpEncodingParameters::max_framerate"
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.

Reason for revert: crbug.com/1028937

Original change's description:
> Add support for RtpEncodingParameters::max_framerate
> 
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
> 
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
2019-11-27 14:01:53 +00:00
Florent Castelli
15be5282e9 Add support for RtpEncodingParameters::max_framerate
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.

Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
2019-11-25 16:43:59 +00:00
Mirko Bonadei
8fa616f3b0 Add 2 more RTC_EXPORT.
The component build test failed close to the end of compilation because
of these two missing symbols, see [1].

[1] - https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/435732

Bug: webrtc:9419
Change-Id: Ic46acf1acbf3bc04e7410f8d9858785739ca98d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159683
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29766}
2019-11-12 09:44:29 +00:00
Mirko Bonadei
54875d05f1 Add missing RTC_EXPORT for the Chromium Windows build.
After fixing the issue with crbug.com/1018579, lld-link complained
that some symbols need to be exported, see [1].

[1] - https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/432025

Bug: webrtc:9419
Change-Id: I9107a9c76361f4c66463a9af2e81a3991ae14df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159007
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29721}
2019-11-07 09:32:24 +00:00
Mirko Bonadei
86d053c2db Use source_sets in component builds and static_library in release builds.
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).

Source sets always pass all the object files to the linker.

On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.

See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set

Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:17:18 +00:00
Erik Språng
f4e0c29ed1 SimulcastEncoderAdapter: support per layer fallback and single encoder proxying
This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.

It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.

Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
2019-10-01 17:31:44 +00:00
Evan Shrubsole
7c079f650d Reland "Fix minor regression caused by a8336d3"
This is a reland of 809198edfff416fce8d75b574a43afab5e67b1cd

A fix was made in https://webrtc-review.googlesource.com/c/src/+/154343
which fixed the regression issues caused by the original patch.

Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}

Bug: webrtc:10126
Change-Id: Iecc3ab6a5cd1193a1fa8e824dcf4f0b8165f9bf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154359
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29356}
2019-10-01 11:49:38 +00:00
Evan Shrubsole
b6a45dda4c Revert "Fix minor regression caused by a8336d3"
This reverts commit 809198edfff416fce8d75b574a43afab5e67b1cd.

Reason for revert: Performance regressions that need to be addressed.

Original change's description:
> Fix minor regression caused by a8336d3
> 
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
> 
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}

TBR=sprang@webrtc.org,eshr@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10126
Change-Id: I133cbe5d8cb894ed944ae8a2d0f63a78bbed72ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153484
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29221}
2019-09-18 14:38:15 +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