This is a reland of commit 09f03be54804e81f626c26e8fde8c86cc952545f
Use max_num_layers instead of encoder_config.number_of_streams when calculation stream resolutions in EncoderStreamFactory::GetStreamResolutions().
Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}
Bug: webrtc:351644568, b/352504711
Change-Id: Ib3fd859257b61c2a5d695b8b8f45c95495117c0e
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42654}
This reverts commit 09f03be54804e81f626c26e8fde8c86cc952545f.
Reason for revert: breaks downstream projects
Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}
Bug: webrtc:351644568, b/352504711
Change-Id: I7aadbe49419b7ac610db4db99284fdcdce9deff5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42653}
a frame must be (or should be) first when it contains either SPS (but not just PPS),
is an IDR or is a slice with first_mb_in_slice == 0.
Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
into a single RTP packet which can happen with small 320x196 frames
BUG=webrtc:352379280,webrtc:346608838
Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42652}
Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
* GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
* GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
* GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
* GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
* GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
* GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
* GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
* GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
Bug: webrtc:351644568, b/352504711
Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42651}
In preparation for upcoming changes in GetSimulcastConfig(), which will require a vector of stream resolutions instead of just the max resolution as an input, switch tests to use CreateEncoderStreams() instead of calling GetSimulcastConfig() directly.
Bug: webrtc:351644568, b/352504711
Change-Id: I541dd54a21a8b75028cff07a250f858a47898223
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357400
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42648}
Merge GetAndSetRtpSendParametersScaleResolutionDownByVP8, GetAndSetRtpSendParametersScaleResolutionDownByVP8WithOddResolution, GetAndSetRtpSendParametersScaleResolutionDownByH264 and GetAndSetRtpSendParametersScaleResolutionDownByH264WithOddResolution into one parameterized test.
PS. Not sure why we need separate tests for VP8 and H264. Underlaying code paths are codec agnostic as I can see.
Bug: webrtc:351644568, b/352504711
Change-Id: Ic95c59c1bfacdba8a42de8ecca9cab42014842e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42646}
All known users are updated to use ntp_time_util.h directly
Bug: webrtc:343076000
Change-Id: I7229b9e5dd72d83bfd98ba4050ae7583d792575b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357300
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42645}
QualityConvergenceController is a layer between VideoStreamEncoder
and QualityConvergenceMonitor that takes care of the simulcast
logic.
Bug: chromium:328598314
Change-Id: Iad8a9d9138e69a60fd508a7ef038220947888f0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356420
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42642}
All calls in code under test were migrated to AudioEncoderFactory::Create and thus there is no longer need to propagate older function.
Bug: webrtc:343086059
Change-Id: I9e0ea4024759deb22c0d284e0e4bac7322a08f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357181
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42638}
This is a cleanup of simulcast.cc. Remove GetNormalSimulcastLayers and GetScreenshareLayers from simulcast.h. Move the implementations to anonymous namespace in simulcast.cc.
Bug: webrtc:351644568, b/352504711
Change-Id: Iff03161e5c44cb0e7faa60b16cfc2fc9b903d5ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357103
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42635}
This is a cleanup of simulcast.cc. max_qp is not needed to decide simulcast config. Move setting of max QP in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams(), where it can be set per stream.
Bug: webrtc:351644568, b/352504711
Change-Id: Ia0e3e9d90032383574dc8867b30d362e9c5df7e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42634}
This is a cleanup of simulcast.cc. bitrate_priority is not needed to decide simulcast config. Move setting of bitrate priority in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams().
Bug: webrtc:351644568
Change-Id: I002d728ccf8d141fe4bbb32b390129ce57c830cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357101
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42629}
num_ref are represented using golomb and may be very large.
BitstreamReader is generally resilent to many consenquites fail reads, but not when number of reads comparable to int limit.
This change address the issue in two ways, either one is enough, but both are helpful in their own way:
H264 parser now fails faster when number of references is too large.
BitstreamReader now is resilent to unlimited number of fail reads.
Bug: chromium:352402499
Change-Id: I19646bc3f53cd2970393d00bc143400b1fdf5473
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357100
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42628}
This reverts commit 4f3d660f4f28a1dcdb577ab89d57830083ac883a.
Reason for revert: downstream still broken
Original change's description:
> Reland "Remove jni_zero type aliases in jni_generator_helper.h."
>
> This is a reland of commit 9fcaa034bc032da9de5d6fcdd45528169f44d343
>
> Original change's description:
> > Remove jni_zero type aliases in jni_generator_helper.h.
> >
> > This CL removes some type alias from
> > sdk/android/src/jni/jni_generator_helper.h and make sure all the
> > jni_zero types are referred to using the jni_zero:: namespace.
> >
> > The goal is to remove sdk/android/src/jni/jni_generator_helper.h
> > in future CLs.
> >
> > Bug: b/319078685, b/351773023
> > Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
> > Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> > Reviewed-by: Jeremy Leconte <jleconte@google.com>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42618}
>
> Bug: b/319078685, b/351773023
> Change-Id: I604b0842d220d76c36b73d2d49bcefe0ee7ae14f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356903
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42626}
Bug: b/319078685, b/351773023
Change-Id: I9429789e44c82735305b2d0d90cbaa4891d79622
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357041
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42627}
This is a reland of commit 9fcaa034bc032da9de5d6fcdd45528169f44d343
Original change's description:
> Remove jni_zero type aliases in jni_generator_helper.h.
>
> This CL removes some type alias from
> sdk/android/src/jni/jni_generator_helper.h and make sure all the
> jni_zero types are referred to using the jni_zero:: namespace.
>
> The goal is to remove sdk/android/src/jni/jni_generator_helper.h
> in future CLs.
>
> Bug: b/319078685, b/351773023
> Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42618}
Bug: b/319078685, b/351773023
Change-Id: I604b0842d220d76c36b73d2d49bcefe0ee7ae14f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356903
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42626}
ScreenCapturerSCK uses some fields that were not available in macOS 13
but the code compiles with the older SDK because of missing annotations
that were added in the macOS 15 SDK.
Bug: chromium:351843815
Change-Id: Ic1a89b4cab43d6ee81d447ccc33ef94439752c45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356860
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42624}
Call it once instead of 3 times.
Also remove FindSimulcastMaxBitrate, FindSimulcastTargetBitrate, FindSimulcastMinBitrate and a one parameter less version of InterpolateSimulcastFormat.
Bug: b/337757868, webrtc:351644568
Change-Id: I7b4002fc3134c47f368bb833925c959a07ce5177
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356980
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42623}
MakeAudioEncoder planned to be removed, and Create planned to become pure virtual
While at it, cleanup nearby mock usage:
Remove ON_CALL that by default return default constructed result
Remove EXPECT_CALL().Times(AnyNumber()) for a NiceMock
Remove parameters in EXPECT_CALL when all are wildcard
Remove redundant get to deference a smart pointer
Bug: webrtc:343086059
Change-Id: Ica90a4980350cb82bcebd11df6c63a01b828bb9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356884
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42622}
Current thresholds were tuned to guarantee no buffer overshoot in an extreme scenario (encoding a high complexity video in a low bitrate).
Bug: b/337757868, webrtc:351644568
Change-Id: I832b2564af6f18f06550338cc9b3618f8acdf831
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356580
Reviewed-by: Dan Tan <dwtan@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42620}
This reverts commit 9fcaa034bc032da9de5d6fcdd45528169f44d343.
Reason for revert: Breaks downstream project.
Original change's description:
> Remove jni_zero type aliases in jni_generator_helper.h.
>
> This CL removes some type alias from
> sdk/android/src/jni/jni_generator_helper.h and make sure all the
> jni_zero types are referred to using the jni_zero:: namespace.
>
> The goal is to remove sdk/android/src/jni/jni_generator_helper.h
> in future CLs.
>
> Bug: b/319078685, b/351773023
> Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
> Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42618}
Bug: b/319078685, b/351773023
Change-Id: I003acf68e2b84bd0a5cda7c7180a28bcd3ca3772
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356902
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42619}
This CL removes some type alias from
sdk/android/src/jni/jni_generator_helper.h and make sure all the
jni_zero types are referred to using the jni_zero:: namespace.
The goal is to remove sdk/android/src/jni/jni_generator_helper.h
in future CLs.
Bug: b/319078685, b/351773023
Change-Id: Ief60fce3e8f301f09ac5392d143aa5a82a445bcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356882
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42618}
The function has been deprecated in favor of
FrameEncodeMetadataWriter::FillMetadataAndTimingInfo().
Bug: chromium:328598314
Change-Id: Iaf2008e855dbd71f2d7cf412d95c5932b3645d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356042
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42613}
This CL adds the sequence checks to the functions that creates or
destroys the encoder context. Those functions must be executed on
the webrtc encoder sequence.
Bug: b/320555128
Change-Id: I1daa93f2f5326073e8d75e1d711d7554bed76a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Hirokazu Honda <hiroh@google.com>
Cr-Commit-Position: refs/heads/main@{#42612}
Similar to BGRA/RGBA we added recently, formats from PipeWire are in
big-endian, while WebRTC (using libyuv) is little-endian, therefore we
have to map BGR to RGB and not RGB to RGB as colors would be off. Also
add some additional formats supported by libyuv.
Bug: webrtc:42225999
Change-Id: Iee8303f0922fe434069b2b3f88994abecf7d2cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355860
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42609}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
