This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ieb6effd55f0ecba17cefc2f07f5eda1e85dbd016
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296441
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39535}
The width/height need to be updated in the VP9 SVC case since resolution
alignment may be applied inside GetSvcConfig(). This is not needed in
the VP9 simulcast case since we don't support simulcast + SVC combo and
resolution alignment is not needed for non-SVC.
This CL gates the "resolution update" behind
"numberOfSimulcastStreams == 1".
Bug: webrtc:14884
Change-Id: Ic3551721dcf6775fea6ff0c85fba48e88069fa5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296766
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39524}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ie6820a820f22635fe7a970db70b9c28d37499848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296443
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39518}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Id1c7fbb969a63eee96fd88c376371aa7eafd0919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296440
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39512}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ieafdb2640b12c254edfac04e98f86f9170c5a71a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295870
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39483}
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread. Avoid this
by posting only if invoked on a diffferent sequence.
TESTED=presubmit + local Meet calls.
Bug: chromium:1373439
Change-Id: I2ca15b2c725f412ca8a0b8132d6b221f9f6b6032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295868
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39477}
This class name better reflects the nomenclature defined by RFC5481: https://datatracker.ietf.org/doc/html/rfc5481#section-1.
Some code style improvements were performed. No functional changes are intended.
Bug: webrtc:14905
Change-Id: I84b9deb7b2ac7f1a07ae00670eaff9656a50c2cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295661
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39466}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ib11fe46f35ab0efba35c6a9a2482b4f7c016226c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295821
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39451}
This reverts commit 8bf3210629.
Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())
Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}
Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
This reverts commit 437bf78ed9.
Reason for revert: Breaks upstream project
Original change's description:
> operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
>
> Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
>
> Also default-initialized VideoFrameMetadata::ssrc_ to 0.
>
> Bug: webrtc:14708
> Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> Commit-Queue: Tove Petersson <tovep@google.com>
> Reviewed-by: Tony Herre <herre@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39411}
Bug: webrtc:14708
Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39413}
Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
Also default-initialized VideoFrameMetadata::ssrc_ to 0.
Bug: webrtc:14708
Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39411}
This can happen when the encoder uses real presentation timestamps that
originate with the input frames. By using those, the encoder can bypass
webrtc frame dropping logic and may severely over/under-shoot if the
timestamps are very precise. In practice, this seems rather common on
Chrome on Windows.
Bug: aomedia:3391
Change-Id: I2be5eed4fabc86dac8a6c7bfdd068c2dcb5a3743
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294740
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39382}
This CL propagates capture_time_identifier introduced in
webrtc::VideoFrame and propagates it to EncodedImage. For use cases
involving EncodedTransforms, this identifier is further propagated to
TransformableVideoSenderFrame.
VideoEncoder::Encode function is overriden by each encoder. Each of
these overriden functions needs to be changed so that they can handle
this new identifier and propagate its value in the created EncodedImage.
Change-Id: I5bea4c5a3fe714f1198e497a4bcb5fd059afe516
Bug: webrtc:14878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291800
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39374}
This uses the field trial introduced is crbug.com/1406331 and
extends the usage to OpenH264. This simplifies experimentation
whether this change improves performance without requiring
multi-slice encoding.
BUG=webrtc:14368
Change-Id: I0031e59059f7113dd5453234869c957d46f311bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39371}
With LTR and SVC etc., H.264 should be able to skip lost frames, and continue to play from the new frames. With DependencyDescriptor, it is allowed to reference the previous frames, even there is a gap in the middle. However, we found there is a special logic for H.264 in packet_buffer.cc, which requires no gap for H.264.
We should allow gaps if the packet has GenericDescriptorInfo (either GenericDescriptor or DependencyDescriptor header extension).
Bug: webrtc:14887
Change-Id: Id66726bab33229bd883f257136ff2e8523fb44c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294062
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39370}
As requested by a CEF hosted application (https://crbug.com/1406331)
who want to be able to limit the number of threads in a controlled
environment, this CL adds a flag to control the max limit per encoder.
For plumbing-reasons, this is placed in VideoEncoder::Settings but
with a note that this is considered an experimental API with limited
support. For now only LibvpxVp8Encoder uses it and there are no plans
to roll this out.
I have manually confirmed this is working with printf debugging,
--force-fieldtrials=WebRTC-VideoEncoderSettings/encoder_thread_limit:2
and https://jsfiddle.net/henbos/2bd6m7Lt/
Bug: chromium:1406331
Change-Id: Ib02bd83e2071034874843d3aaa0d3b0adc5bbf46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293960
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39349}
The CodecTimer is not a codec timer, it's more like a decoder stopwatch with a percentile filter wrapped around it. Since the purpose of the class is to provide an estimate for how much decode delay to add when determining the render timestamp of a frame, let's rename this class to `DecodeTimePercentileFilter`.
No functional changes are intended.
Bug: webrtc:14905
Change-Id: I48c99e4f500c4f9e1a2a20b0afe72d6e76c5192d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293462
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39332}
The kMinimumFrameRate constant is only used in a comparison with
RateControlParameters::framerate_fps, which is of the double type.
Declare kMinimumFrameRate as double to match.
Note: The kMinimumFrameRate constant was added in
https://webrtc-review.googlesource.com/c/src/+/170360.
Bug: webrtc:11404
Change-Id: I11769867d4e52a720219c8a0ade8e8b74d13ca86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293384
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#39320}
As part of go/unblocking-vp9-simulcast (Step 1), EncodedImage is being
upgraded to be able to differentiate between what is a simulcast index
and what is a spatial index.
In order not to break existing code assuming that "if codec != VP9,
SpatialIndex() is the simulcast index", SimulcastIndex() has fallback
logic to return the value of spatial_index_ in the event that
SetSimulcastIndex() has not been called. This allows migrating external
code from (Set)SpatialIndex() to (Set)SimulcastIndex(). During this
intermediate time, codec gates are still necessary in some places of
the code, see TODOs added.
In a follow-up CL, after having fixed dependencies, we'll be able to
remove the fallback logic and rely on SimulcastIndex() and
SpatialIndex() actually being the advertised index and "if codec..."
hacks will be a thing of the past!
Bug: webrtc:14884
Change-Id: I70095c091d0ce2336640451150888a3c3841df80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293343
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39318}
PostDelayedTask doesn't guarantee task execution order. For example,
if you post two tasks, A and B, back-to-back using the same delay
there is no guarantee that A will be executed before B.
Re-implemented pacing using sleep(). Changed pacer to compute task
scheduled time instead of delay. Sleep time is calculated right before
task start. This provides better accuracy by accounting for any delays
that may happen after pacing time is computed and before task queue is
ready to run the task.
It is tricky to implement pacer tests using simulated clocks. The test
use system time which make them flacky on low performance bots. Keep
the test disabled by default.
Bug: b/261160916, webrtc:14852
Change-Id: I88e1a2001e6d33cf3bb7fe16730ec28abf90acc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291804
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39302}
The elements of the fps_allocation vector are fractions of the maximum
frame rate. Each fraction is represented as an 8-bit unsigned integer,
where 0 = 0% and 255 = 100%.
The original code (added in
https://webrtc-review.googlesource.com/c/src/+/201384) sets the elements
of the fps_allocation vector to frame rates rather than frame rate
fractions. Perhaps fps_allocation could be renamed to avoid this kind of
confusion.
modules_unittests --gtest_filter=LibaomAv1EncoderTest.*
Tested:
Change-Id: Icd050da3b3c2cff31913c3430f7b6b6e9829b9fa
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292784
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39286}
Passing of ownership of codecs to tester is not strictly needed. We may need to continue using a codec after test. For example, to check codec state or to use the same codec instance in next test.
Bug: b/261160916, webrtc:14852
Change-Id: I179b262116d7de76b8171f0409f943ad6d87433e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291802
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39256}
Step 1 of combining the sender and receiver types
Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used
by the transformable receiver frame, I couldn't work out a better way
of managing the dependencies, and everything else seemed to work fine.
Bug: chromium:1412687
Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39255}
This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters.
VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl.
Bug: b/261160916, webrtc:14852
Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39248}
Before this, an empty list of CSRCs was always provided up to encoded
insertable streams transforms for remote video tracks, regardless of
the actual CSRCs on received frames. Audio already works correctly.
Bug: chromium:1411614
Change-Id: I51ab4dc5e67a1a35893fefff16c1f057e9047e6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291539
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39240}
This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate.
Bug: webrtc:14818
Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39123}
Fallback to a default value if the scalability mode is unset or not supported by the codec.
The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set).
Bug: webrtc:11607
Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39074}
We found that the legacy assumption for H264 which assumed that
simulcast streams would use 2x width ratios in unnecessary as the
encoder has since been fixed to handle multiple ratios.
H264 encoder still works even if this assumption is invalid
Bug: None
Change-Id: I9caacf78d26c8215b94858a2d8674ec4cd64e96e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286940
Reviewed-by: Mirta Dvornicic <mirtad@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39072}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.
Test failures seem unrelated, so using No-Try.
No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
This tester is an improved version of VideoProcessor and VideoCodecTestFixture and will eventually replace them.
The tester provides better separation between codecs and testing logic. Its knowledge about codecs is limited to frame encode/decode calls and frame ready callbacks. Instantiation and configuration of codecs are the test responsibilities.
Other differences:
- Run encoding and decoding in separate threads
- Run quality analysis in a separate thread
- Reference frame buffering is moved into video source (which re-read frames from the file).
- Make it possible to run decode-only tests
This CL is MVP implementation: it adds only 1 test (video_codec_test.cc, ConstantRate/EncodeDecodeTest) and the test is disabled for now.
Bug: b/261160916
Change-Id: Ida24a2fca1b1496237fa695c812084877c76379f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283525
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38901}
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.
Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
