Commit graph

318 commits

Author SHA1 Message Date
Niels Möller
13d163654a Delete support for has_internal_source
Bug: webrtc:12875
Change-Id: I9683e71e1fe5b24802033ffcb32a531ca685fc6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179220
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35353}
2021-11-16 11:29:40 +00:00
Artem Titov
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
Harald Alvestrand
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
2021-11-04 13:46:17 +00:00
Emil Lundmark
7194d832b2 Make AV1X constants private
The constants are being made private since no new code should use them.
However, the helper functions sill uses "AV1X" internally for backwards
compatibility.

Bug: webrtc:13166
Change-Id: I0a0cd46f31ca70bb7f395c9b1e9cdb202df11f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35289}
2021-11-01 09:48:50 +00:00
Jiwon Jung
3791d332c5 Proper assignment of FEC setting to encoders in SoftwareFallbackWrapper.
When using VideoEncoderSoftwareFallbackWrapper, releasing and
initialization of encoder_ (H/W) and fallback_encoder_(S/W) happen
repeatedly as reconfiguration procedure is called from higher layer.

Below problems would occur when our encoder_(H/W) fails to initialize
or encode.

Firstly, some encoders' SetFecControllerOverride() functions will fail
during repeated calls since they have checks like
RTC_DCHECK(!fec_controller_override_) to avoid repeated assignment of
fec_controller_override_.
(see : LibvpxVp8Encoder::SetFecControllerOverride())

Secondly, if main_ encoder fails to initialize at first attempt, FEC
setting (fec_controller_override) will not set until reconfiguration
procedure is called again.

This CL comes with two changes to fix above problems.
1. Sets fec_controller_override to both encoders when
SoftwareFallbackWrapper::SetFecController() is called.
2. Removes the current_encoder()->SetFecControllerOverride() in
PrimeEncoder() to avoid redundant calls which may involve fatal error.

Bug: webrtc:13184
Change-Id: I082c93de552bc9ec3141c6490d35acfcee2f8935
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234301
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35231}
2021-10-19 09:56:12 +00:00
Ilya Nikolaevskiy
54f377308f Revert "Added support for H264 YUV444 (I444) decoding."
This reverts commit 7d8ed34372.

Reason for revert: Breaks internal builds

Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> Bug: chromium:1251096
> Change-Id: Ib47c2b1f94797afb6c5090f3c46eae6f13110992
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234540
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35200}

TBR=ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,peterhanspers@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com,stefan.mitic@nutanix.com

Change-Id: I3048c353a2b6b4f3d4e5e53a88f48b456f1ce593
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1251096
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235203
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35207}
2021-10-14 12:59:29 +00:00
Stefan Mitic
7d8ed34372 Added support for H264 YUV444 (I444) decoding.
Added Nutanix Inc. to the AUTHORS file.

Bug: chromium:1251096
Change-Id: Ib47c2b1f94797afb6c5090f3c46eae6f13110992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234540
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35200}
2021-10-14 11:06:55 +00:00
Sergey Silkin
6b19d8273b Replace AV1X with AV1
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".

Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
2021-09-14 08:29:02 +00:00
Danil Chapovalov
78a8ce0a4c Delete decoder specific buffer_pool_size from webrtc::VideoCodec
VideoDecoder no longer uses this VideoCodec class,
thus this member is unused.

Bug: webrtc:13045
Change-Id: I6e46a563e90f2538bf288995a3837d95c00ba9cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230941
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34896}
2021-09-01 15:20:31 +00:00
Danil Chapovalov
00dd5ced24 Delete deprecated VideoDecoder::InitDecode
Bug: webrtc:13045
Change-Id: Id1ca822c3be5a4f496dd67b59eab31a79a74bf67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228949
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34884}
2021-08-31 14:13:49 +00:00
Johannes Kron
3cd7a0ffdd Remove media/base/h264_profile_level_id.* and media/base/vp9_profile.h
The content of these files was moved to api/video_codecs in
https://webrtc.googlesource.com/src.git/+/c3fcee7c3a7714afc3e37d4753b40f4fdbc3653e
but the original files could not be removed due to dependencies
in downstream projects.

Bug: chromium:1187565
Change-Id: I414efa22102bfdea0765fa72a8cf8b0bd5c090db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34869}
2021-08-30 10:31:08 +00:00
Johannes Kron
9ad972d4fb Remove deprecated signature of VideoDecoderFactory::QueryCodecSupport
This function was deprecated in this CL
https://webrtc-review.googlesource.com/c/src/+/229184

Bug: chromium:1187565
Change-Id: Ic0e18af69185b48accc441c4bbe1a2d8926db383
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230241
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34868}
2021-08-30 10:26:05 +00:00
Johannes Kron
3bb74f3800 Change VideoDecoderFactory::QueryCodecSupport to use reference_scaling
All decoders are supposed to be able to decode all valid bitstreams
that can be produced by an encoder. In the cases where this is not
the case, reference_scaling better captures the cause of this than
scalability_mode which was used initially.

Bug: chromium:1187565
Change-Id: I21174077badf0fb9d90b1b58f003edac5b8ee0f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229184
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34800}
2021-08-19 10:21:16 +00:00
Danil Chapovalov
ba0a306585 Move check for number_of_cores parameter validitity
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings

Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
2021-08-14 11:51:53 +00:00
Danil Chapovalov
7fa3f40626 Migrate software fallback wrapper to new VideoDecoder::Configure
Bug: webrtc:13045
Change-Id: I5082a5d12a43313842f8d5eb1fa70a12671e572c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228434
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34733}
2021-08-12 10:21:19 +00:00
Artem Titov
cfea2182f8 Use backticks not vertical bars to denote variables in comments
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +00:00
Danil Chapovalov
ecc46eff5b Introduce new api to initialize VideoDecoder
Bug: webrtc:13045
Change-Id: If14fa3998176ee07b6f2835745568f70347ccac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227766
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34694}
2021-08-10 08:42:43 +00:00
Qiu Jianlin
b54cfdebfe Add optional is_qp_trusted property for EncoderInfo.
Some hardware H.264 encoders does not place average QP delta in
slice_qp_delta field. Adding an optional flag in EncoderInfo to notify
quality scaler about this.

Bug: webrtc:12942
Change-Id: I3ee29c5ae9bd7bb34d26eba7e6bede3798ca44b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34627}
2021-08-02 13:49:21 +00:00
Artem Titov
0e61fdd27c Use backticks not vertical bars to denote variables in comments for /api
Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
2021-07-26 18:27:34 +00:00
Ilya Nikolaevskiy
c98aebbbef Change how alignment requirements are processed
Software fallback wrapper now reports least common multiple of requirements
for two encoders.

SimulcastEncoderAdapter queries actual encoder before InitEncode call
and requests alignment for all layers if simulcast is not supported by
any of the encoders.

Bug: chromium:1084702
Change-Id: Iaed8190737125d447036b6c664b863be72556a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225881
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34466}
2021-07-13 16:49:13 +00:00
Peter Kasting
286b1db1b2 Fix -Wunreachable-code-aggressive.
Bug: chromium:1066980
Change-Id: I6888ea1fbc458c9b3063b3f60a7732af16ab5fc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224266
Reviewed-by: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34393}
2021-06-30 11:14:37 +00:00
Erik Språng
75941453c5 Fix incorrect fps_allocation printed by EncoderInfo::ToString()
Bug: webrtc:12788
Change-Id: I6188a4041265a16f30d37806e555dcb3210cb653
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220927
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34185}
2021-06-01 09:53:34 +00:00
philipel
dab3ce8f29 Reland "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
This reverts commit 49c293f03d.

Reason for revert: Downstream updated

Original change's description:
> Revert "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
>
> This reverts commit 4ba1044bae.
>
> 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}

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

Bug: webrtc:9106
Change-Id: I52923c0f3952832c50a7d73b466775b8c5d541cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216329
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33884}
2021-04-30 11:40:38 +00:00
Johannes Kron
c29e1f5415 Add QueryCodecSupport() to VideoEncoderFactory and VideoDecoderFactory
QueryCodecSupport() is needed in the coming work with extending
the MediaCapabilities API to WebRTC. In the API, a user can query
whether a specified media configuration is supported, power
efficient, and smooth. QueryCodecSupport() will be used to determine
if the configuration is supported and power efficient.

Bug: chromium:1187565
Change-Id: Ib1d93433a180b433f0bf60d6e871d03cebc4c0a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215922
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33843}
2021-04-27 10:23:05 +00:00
Johannes Kron
20ee02c49f Add codec comparison function to SdpVideoFormat
SdpVideoFormat is used to configure video encoder and decoders.
This CL adds support for comparing two SdpVideoFormat objects
to determine if they specify the same video codec.

This functionality previously only existed in media/base/codec.h
which made the code sensitive to circular dependencies. Once
downstream projects stop using cricket::IsSameCodec, this code
can be removed.

Bug: chromium:1187565
Change-Id: I242069aa6af07917637384c80ee4820887defc7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213427
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33794}
2021-04-21 07:21:41 +00:00
Johannes Kron
c3fcee7c3a Move h264_profile_level_id and vp9_profile to api/video_codecs
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.

The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.

Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-20 09:42:05 +00:00
Sergey Silkin
d19e3b9676 Reland "Reland "Enable quality scaling when allowed""
This reverts commit 31c5c9da35.

Reason for revert: made QP parser thread-safe https://webrtc.googlesource.com/src/+/0e42cf703bd111fde235d06d08b02d3a7b02c727

Original change's description:
> Revert "Reland "Enable quality scaling when allowed""
>
> This reverts commit 0021fe7793.
>
> Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview
>
> Original change's description:
> > Reland "Enable quality scaling when allowed"
> >
> > This reverts commit eb449a979b.
> >
> > Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
> >
> > Original change's description:
> > Before this CL quality scaling was conditioned on scaling settings
> > provided by encoder. That should not be a requirement since encoder
> > may not be aware of quality scaling which is a WebRTC feature. In M90
> > chromium HW encoders do not provide scaling settings (chromium:1179020).
> > The default scaling settings provided by these encoders are not correct
> > (b/181537172).
> >
> > This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> > is set to true in singlecast with normal video feed (not screen sharing)
> > mode. If quality scaling is allowed it is enabled no matter whether
> > scaling settings are present in encoder info or not. Setting from
> > QualityScalingExperiment are used in case if not provided by encoder.
> >
> > Bug: chromium:1179020
> > Bug: webrtc:12511
> > Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33438}
>
> TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com
>
> Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1179020
> Bug: webrtc:12511
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33439}

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

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I3a31e1c1fdf7da93226f8c1e0a96b43fe0b786df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212026
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33481}
2021-03-16 15:13:52 +00:00
Ilya Nikolaevskiy
31c5c9da35 Revert "Reland "Enable quality scaling when allowed""
This reverts commit 0021fe7793.

Reason for revert: Broken on linux_tsan bot: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview

Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit eb449a979b.
>
> Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33438}

TBR=brandtr@webrtc.org,ilnik@webrtc.org,ssilkin@webrtc.org,rubber-stamper@appspot.gserviceaccount.com

Change-Id: Id7633a1e98f95762e81487887f83a0c35f89195c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1179020
Bug: webrtc:12511
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211352
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33439}
2021-03-11 15:14:42 +00:00
Sergey Silkin
0021fe7793 Reland "Enable quality scaling when allowed"
This reverts commit eb449a979b.

Reason for revert: Added QP parsing in https://webrtc.googlesource.com/src/+/8639673f0c098efc294a7593fa3bd98e28ab7508

Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I97911fde9005ec25028a640a3f007d12f2bbc2e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211349
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33438}
2021-03-11 13:43:11 +00:00
Guido Urdaneta
eb449a979b Revert "Reland "Enable quality scaling when allowed""
This reverts commit 83be84bb74.

Reason for revert: Suspect of crbug.com/1185276

Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33385}

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33393}
2021-03-06 09:40:50 +00:00
Sergey Silkin
83be84bb74 Reland "Enable quality scaling when allowed"
This reverts commit 609b524dd3.

Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.

Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33385}
2021-03-04 16:01:23 +00:00
Björn Terelius
609b524dd3 Revert "Enable quality scaling when allowed"
This reverts commit 752cbaba90.

Reason for revert: The test VideoStreamEncoderTest.QualityScalingAllowed_QualityScalingEnabled seems to fail on iOS.

Original change's description:
> Enable quality scaling when allowed
>
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020, webrtc:12511
> Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33373}

Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Icabf2d9a034d359f79491f2c37f1044f17a7445d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209641
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33381}
2021-03-04 10:11:36 +00:00
Sergey Silkin
752cbaba90 Enable quality scaling when allowed
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).

This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.

Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33373}
2021-03-03 13:57:22 +00:00
Danil Chapovalov
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.

Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
2021-02-22 12:53:23 +00:00
Åsa Persson
aab91c7b3a Remove temporal layer restriction for forced resolution based fallback.
Bug: none
Change-Id: Id8d30b6759bc6d5a517d1363395d4495251b32fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205860
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33172}
2021-02-05 07:31:54 +00:00
philipel
25b8235f03 Remove unused function VideoDecoder::PrefersLateDecoding.
Bug: webrtc:12271
Change-Id: Iaf67df37c0eade8b0b6f38be122530c3d908cf35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201820
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33028}
2021-01-18 14:17:57 +00:00
Erik Språng
c12f625938 Adds VideoDecoder::GetDecoderInfo()
This adds a new way to poll decoder metadata.
A default implementation still delegates to the old methods.
Root call site is updates to not use the olds methods.

Follow-ups will dismantle usage of the olds methods in wrappers.

Bug: webrtc:12271
Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32976}
2021-01-14 13:33:22 +00:00
philipel
360da05ed1 Remove webrtc::VideoDecoder::PrefersLateDecoding.
This is just general cleanup.

The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).

Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
2021-01-11 18:02:25 +00:00
Niels Möller
08d2c2bf46 Delete unneeded dependencies on the Module abstraction
Bug: webrtc:7219
Change-Id: I1bcbab7e30f9964798a093e888b07d758cf226e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198124
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32865}
2020-12-21 09:09:57 +00:00
philipel
87e99095a7 Make video scalability mode configurable from peerconnection level.
This CL does not aim at cleaning up simulcast/SVC configuration, just to make it possible to set the scalability mode for AV1. Implementing a codec agnostic SVC/simulcast API is a (big) project on its own.

Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca

BUG: webrtc:11607
Change-Id: Ia88df31eb1111713e5f8832e95c8db44f92887ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192541
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32631}
2020-11-18 12:06:03 +00:00
Jakob Ivarsson
a9961b3839 Allow temporal based switch if temporal layers are undefined.
Bug: webrtc:11324
Change-Id: Iee4717f453bb9883683d752832fbc7bf999a96c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193704
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32630}
2020-11-18 10:59:22 +00:00
Karl Wiberg
c95b939667 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.

Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
2020-11-09 10:47:55 +00:00
Fabien Vallée
f8b5bfeaf2 Fix "control reaches end of non-void function" warnings
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)

In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.

Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
2020-10-27 10:22:23 +00:00
Danil Chapovalov
9f4859e5e3 Allow to set av1 scalability mode after encoder is constructed
Bug: webrtc:11404
Change-Id: I70b4115c8afdc4f32fd876d31d54b7d95d0a7e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188582
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32437}
2020-10-19 10:42:23 +00:00
Evan Shrubsole
5089a8ea14 Use VideoFrameBuffer::Scale in encoder wrappers
This sincludes the SimulcastEncoderAdapter and the
VideoEncoderSoftwareFallbackWrapper. This avoids converting
the frame when that is not needed.

Bug: webrtc:11976
Change-Id: I686725ecfb79c3b8d87d587a907da1602483bfe8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32389}
2020-10-13 12:33:42 +00:00
Ilya Nikolaevskiy
38e9b06151 Reland "Add scaling interface to VideoFrameBuffer"
(Reland with no changes after the fix to the downstream project)

This can be overriden for kNative frame types to perform scaling efficiently.

Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.

Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303

(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org

Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
2020-10-09 08:30:50 +00:00
Ilya Nikolaevskiy
441dbf9a56 Revert "Add scaling interface to VideoFrameBuffer"
This reverts commit c79f1d8cfb.

Reason for revert: Breaks downstream project.

Original change's description:
> Add scaling interface to VideoFrameBuffer
>
> This can be overriden for kNative frame types to perform scaling efficiently.
>
> Default implementations for existing buffer types require actual
> buffer implementation, thus this CL also merges "video_frame"
> with "video_frame_I420" build targets.
>
> Bug: webrtc:11976, chromium:1132299
> Change-Id: I3bf5f6bf179db5e7ab165b1c2301980043a08765
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#32352}

TBR=mbonadei@webrtc.org,sakal@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,eshr@google.com

Change-Id: I86ac697bf963ef7e2c4f2ed34c3a7bf04f4f1ce1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11976
Bug: chromium:1132299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187344
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32354}
2020-10-08 14:16:23 +00:00
Evan Shrubsole
b556b08668 Allow encoders to receive preferred pixel formats from native buffers
Adds a field to EncoderInfo called preferred_pixel_formats which a
software encoder populates with the pixel formats it supports. When a
kNative frame is received for encoding, the VideoStreamEncoder will
first try to get a frame that is accessible by the software encoder in
that pixel format from the kNative frame. If this fails it will fallback
to converting the frame using ToI420.

This minimizes the number of conversions made in the case that the
encoder supports the pixel format of the native buffer or where
conversion can be accelerated. For example, in Chromium, the capturer can
emit an NV12 frame, which can be consumed by libvpx which supports NV12.

Testing: Tested in Chrome with media::VideoFrame adapters.

Bug: webrtc:11977
Change-Id: I9becc4100136b0c0128f4fa06dedf9ee4dc62f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32353}
2020-10-08 13:47:50 +00:00
Ilya Nikolaevskiy
c79f1d8cfb Add scaling interface to VideoFrameBuffer
This can be overriden for kNative frame types to perform scaling efficiently.

Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.

Bug: webrtc:11976, chromium:1132299
Change-Id: I3bf5f6bf179db5e7ab165b1c2301980043a08765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#32352}
2020-10-08 13:33:00 +00:00
Per Kjellander
d0a8f51ef7 Reland Refactor reporting of VideoBitrateAllocation
Original description
 Move reporting of target bitrate to just after the encoder has been
 updated. Originall submitted as refs/heads/master@{#32275}

Patch 1 contains the original cl
,patch 2 the fix to send rtcp even if BWE does not change.

Bug: webrtc:12000
Change-Id: I16766e08229fe1f6f65f449e0e074bed03338693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186948
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32340}
2020-10-07 11:06:56 +00:00
Per Kjellander
b8781796d4 Revert "Refactor reporting of VideoBitrateAllocation"
This reverts commit 39a31afb77.

Reason for revert: Will cause RTCP Target bitrate messages to not be sent unless BWE changes. 

Original change's description:
> Refactor reporting of VideoBitrateAllocation
>
> Move reporting of target bitrate to just after the encoder has been
> updated.
>
> Bug: webrtc:12000
> Change-Id: I3e7c5bd44c2f64e5f7e32d6451861b80e0b779ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186041
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32275}

TBR=sprang@webrtc.org,perkj@webrtc.org

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

Bug: webrtc:12000
Change-Id: Icf21e6ae28dc17c61b9243c037ffef9b623894ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186945
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32337}
2020-10-07 08:14:21 +00:00
Evan Shrubsole
895556e19c Avoid converting frames to I420 in VideoStreamEncoder
This needs to be done still for kNative frames, but all other frame types
can be passed in.

I have checked all VideoEncoder implementations in Chromium and confirmed they either convert the frame to their preferred pixel format, or just
forward the frame to a delegate encoder.

Tested:
- video_loopback with NV12 generated frames for VP9, the only
codec supporting NV12, as well as VP8 which only accepts I420 frames.
- internal_tests tryrun

Bug: webrtc:11976,webrtc:11635
Change-Id: If39a815fb0c5636fceb1040c8946c3db2fb350a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185803
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32306}
2020-10-05 08:33:46 +00:00
Per Kjellander
39a31afb77 Refactor reporting of VideoBitrateAllocation
Move reporting of target bitrate to just after the encoder has been
updated.

Bug: webrtc:12000
Change-Id: I3e7c5bd44c2f64e5f7e32d6451861b80e0b779ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186041
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32275}
2020-10-01 14:10:09 +00:00
Åsa Persson
c5a74ffba4 Add support so requested resolution alignment also apply to scaled layers.
Bug: webrtc:11872
Change-Id: I7f904e2765330ee93270b66b0102ce57f336f9a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181883
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32146}
2020-09-21 09:23:22 +00:00
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 fdd41ecf6b

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 32ca95145c

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 fdd41ecf6b.

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 32ca95145c.

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 4ba1044bae.

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 6958d2c6f0.

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 3a2be87b80.

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 15be5282e9.

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