This reverts commit 88fe84b7fb.
Reason for revert: Downstream project isn't updated to the latest libvpx roll yet, thus some tests are broken.
Original change's description:
> VP9 encoder: handle disabled layers correctly
>
> Now vp9 screenshare would enable new layers as soon as requested and will
> force all spatial layers present on the next frame, even if they should be
> dropped because of frame-rate limiting.
>
> This might cause frame-rate liming to be exceeded if layer is toggling on
> and off very often, but this situation is bad itself. E.g. in realtime video
> it will cause too many key-frames.
>
> Now SvcRateAllocator and VP9EncoderImpl are aware that there may be some skipped
> layers before the first enabled. Key-frames and ss_info triggering logic is also
> updated.
>
> Bug: webrtc:10977
> Change-Id: Ie2555210c0368a1d3c51ddf6670d0052e6d679de
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29296}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
Change-Id: If33886a5f8a0c3b33168dcadfe45c11a6f4387c1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10977
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154354
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29299}
Now vp9 screenshare would enable new layers as soon as requested and will
force all spatial layers present on the next frame, even if they should be
dropped because of frame-rate limiting.
This might cause frame-rate liming to be exceeded if layer is toggling on
and off very often, but this situation is bad itself. E.g. in realtime video
it will cause too many key-frames.
Now SvcRateAllocator and VP9EncoderImpl are aware that there may be some skipped
layers before the first enabled. Key-frames and ss_info triggering logic is also
updated.
Bug: webrtc:10977
Change-Id: Ie2555210c0368a1d3c51ddf6670d0052e6d679de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29296}
Passing an empty arg is working at the moment but it is not
guaranteed to continue to work in the future.
This CL has been generated with:
git grep -l "INSTANTIATE_TEST_SUITE_P(," | xargs sed -i \
"s/INSTANTIATE_TEST_SUITE_P(,/INSTANTIATE_TEST_SUITE_P(All,/g"
Bug: None
Change-Id: Icd2fb9d9d29aed5d692a234124bd990d0f097db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153890
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29282}
This mode was added by libvpx team specificaly for this usecase: if a
layer is dropped, all lower layers have to be dropped also.
This ensures that higher layers always have higher framerate than the
lower layers and stream is RTP compatible.
This CL also renames full_superframe_drop_ to !layer_buffering, as it
closer reflects the purpose of that flag (in screenshare mode, no
buffering is needed, because the highest layer is always present in the
superframe, yet, it's not a full-superframe dropping mode).
Bug: webrtc:10257
Change-Id: I2589bfd2b9b63de0e410f277a716276234993843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151764
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29155}
if higher layer is enabled, then disabled, then key-frame is issued, then
the layer is enabled again, the buffer would contain a picture from before
the key-frame and it might have a higher pid than the currently encoded one.
This would trigger the DCHECK. It's safe to remove the DCHECK completely, because
such occasions would cause unsigned overflow and cause the following check for
maximum allowed picture difference to fail and the wrong picture won't
be used as a temporal reference.
This error only caused failures in debug builds and couldn't lead to corruptions
because there're periodical key-frames generated and pid difference can never become so
big that negative value would overflow to something close to 0.
Bug: webrtc:10257
Change-Id: Ie3b3ed0e24421787e3b40a37987ccecb75d04635
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151643
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29099}
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.
Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
This prepares for using VideoFrameBuffer::Type as
FrameGenerator::OutputType, which will reduce the
number of redundant enums in the code.
Bug: webrtc:9883
Change-Id: I253f5f1ea7181e02a5cf1a92925f51da8ada6aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28696}
This partially reverts these 2 CLs:
1) Reland "Copy video frames metadata between encoded and plain frames in one place"
https://webrtc.googlesource.com/src/+/2ebf5239782bf6b46d4aa812f34fa9f9e5a02be9
2) Don't copy video frame metadata in each encoder/decoder
https://webrtc.googlesource.com/src/+/ab62b2ee51e622be6d0aade15e87e927fa60e6f2
The problem with them were that ColorSpace was made to always be copied from the
EncodedImage in the GenericDecoder, which overwrote ColorSpace information from
the decoder.
If decoder applied color space transition or bitstream color space information
was different from the WebRTC signaled one, the incorrect color space data were
passed to the renderer.
This CL removes introduced change regarding color space data: GenericDecoder
doesn't copy or store it and software decoders are restored to copy it.
Relevant tests are also removed.
Bug: chromium:982486
Change-Id: I989e01476ff7f7df376c05578ab8f540b95a1dd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145323
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28556}
- Don't reset encoder if max/min bitrate changed.
- Removed min/max bitrate DCHECKs from encoder wrappers.
- Reset encoder if start_bitrate changed. Only do this if encoding
has not yet started.
- Updated ReconfigureBitratesSetsEncoderBitratesCorrectly test.
- Removed EncoderSetupPropagatesCommonEncoderConfigValues test since it
was a subset of ReconfigureBitratesSetsEncoderBitratesCorrectly.
Bug: webrtc:10773
Change-Id: Id9cbb2ea229232fd95967819e2a937b26948de9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144028
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28446}
Previously, FecControllerOverride was passed to
Vp8FrameBufferController::SetFecControllerOverride. Passing to
the factory is a more elegant way, since it's only used when
the controller is constructed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28443}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
Rename structures to match terminology in the spec
Bug: webrtc:10342
Change-Id: I1329abaca98ae7f82307451032d5ce1533e80772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28402}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
This is a reland of 11dfff0878
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
As this is handled higher up the pipeline in a single
place for all encoders/decoders
Bug: webrtc:10460
Change-Id: I95b0a69aecaf07283c8776ac0d7e85d097e3576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139882
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28172}
Using this flag, an encoder may inform the RTP sender module that
the packet is not elligible for retransmission. Specifically, it
may not be retransmitted in response to a NACK message,
nor because of early loss detection (see CL #135881).
Bug: webrtc:10702
Change-Id: Ib6a9cc361cf10ea7214cf672e05940c27899a6be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140105
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28169}
Because of a low bitrate target, base layer has drops much more frequently
than other layers. But it reduces overall framerate, especially then
input framerate is low (5 fps).
This CL allows pre-layer drops and disables droppoing on higher spatial
layers for screenshare, solving the issue.
Additional care have to be taken then new spatial layers are enabled
dynamically to not create non-compatible with RTP references.
Bug: webrtc:10257
Change-Id: Ie056484c99a3f35ff4405ef71337dc2d034db8bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138262
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28063}
This CL makes it more flexible and easier to include/exclude H264 code
when using other build systems because it delegates the decision to
remove the code to the preprocessor instead of GN.
This CL should be a noop, and for WebRTC/Chromium the GN param
`rtc_use_h264` will still be the only thing to change in order to
include/exclude H264.
Moving code that requires ffmpeg or h264 out of the #ifdef/#endif
part should break the build since dependencies are only added if
`rtc_use_h264=true`.
Bug: webrtc:9213
Change-Id: Ibc04edc2f6b9e51489ffe638d5be4b32959cdca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137430
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28055}
Reland with fixes.
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: Id40a780d461e6b51cb44d275b8aa5d7b348d3586
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138215
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28054}
This reverts commit eb1754c575.
Reason for revert: breaks downstream projects
Original change's description:
> VP9 screenshare: Don't base layers frame-rate on input frame-rate
>
> If input framerate is a little unstable, using it to cap layers will
> make output framerate even smaller for longer periods of time.
>
> Also, fix screenshare_loopback test for low-fps vp9 testing.
>
> Bug: webrtc:10257
> Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28050}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
Change-Id: I82bfbac58249cfe0da5ff565aa97a4745fd078ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138213
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28051}
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28050}
Ensure that frame_buffer_controller_ does not get assigned null
by the factory.
Bug: None
Change-Id: I84e141ae0390cd024863f88cdcdc79b8b13e7c64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137043
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27962}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
This reverts commit bd20c3f5ae.
Reason for revert: chromium:961253
This CL is not the cause of the regression, but reverting it will make the reverting of the actual cause easier.
Original change's description:
> Rename configurations_ to vpx_configs_ in LibvpxVp8Encoder
>
> Bug: None
> Change-Id: I548a724f0fb81f46785517c90e527edc075e1476
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135040
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27836}
TBR=brandtr@webrtc.org,eladalon@webrtc.org
Bug: chromium:961253
Change-Id: I707337e0ce50f29f9cda7cf45500c11debace1a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135750
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27898}
This reverts commit 4fb12b0cae.
Reason for revert: Breaks some asan chromium bots
Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
>
> Reland with fixes.
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
>
> Bug: webrtc:10460
> Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27828}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10460
Change-Id: I9c87a43a716622b389974cb8377f973573fc29a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135747
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27895}
Make Vp8FrameBufferController::UpdateConfiguration return a set
of desired overrides. These overrides are cumulative with
previously returned override sets.
Bug: webrtc:10382
Change-Id: I1aa9544ae0cf6c57115e80963b3bbcdc3101db5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134649
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27835}
Reland with fixes.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27828}
The color space can either be specified in the VUI of the H264 bitstream
or using an RTP header extension. The color space set through the RTP
header extension overrides the color space in the VUI. The check for
HDR should look at the resulting color space.
Bug: webrtc:10575
Change-Id: I0ca6262d76d56dea938de169f55ad3894e6c4f8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134860
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27816}
In this CL:
- Assign frame IDs so that simulcast streams share one frame ID space.
- Added a CodecBufferUsage class that represent how a particular buffer
was used (updated, referenced or both).
- Calculate frame dependencies based on the CodecBufferUsage information.
Bug: webrtc:10342
Change-Id: I4ed5ad703f9376a7d995c04bb757c7d214865ddb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131287
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27784}
8-bit H264 HDR content is not rendered correctly in Chrome on Windows.
This is a temporary fix that converts the 8-bit buffer to a 10-bit
buffer if the color space indicates that the buffer should be
rendered as HDR.
Bug: webrtc:10575
Change-Id: I106612ec489c6371fa774424a4cf07d9bad40fc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134040
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27766}
This reverts commit c9a2c5e93a.
Reason for revert: Breaks downstream test
Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
>
> Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
>
> Bug: webrtc:10460
> Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27756}
TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org
Change-Id: I34cc563ec6383735c2a76a6f45a72a7726b74421
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134204
Reviewed-by: Artem Titarenko <artit@google.com>
Commit-Queue: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27765}
Reland with fixes: Do not remove extra metadata copies in software decoders as some downstream projects assumes these fields are copied by the encoders.
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Bug: webrtc:10460
Change-Id: I8e49589bf75f406e2b5ddee34882de0faedbd09a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27756}
LibvpxVp8Encoder::Encode() creates a local instance of
rtc::scoped_refptr<I420BufferInterface>, then sets members to
point into the internal state of that I420BufferInterface. These
pointers remain in place after the buffer is destroyed.
This CL fixes the issue by deleting the references when the
function exits.
Bug: webrtc:10570
Change-Id: I9623e2ff3dd43e8fd1d1cc7696a3f28227d4544b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133882
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27738}
This reverts commit 00d0a0a1a9.
Reason for revert: Breaks downstream tests
Original change's description:
> Copy video frames metadata between encoded and plain frames in one place
>
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
>
> Also, added some missing tests.
>
> Bug: webrtc:10460
> Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27719}
TBR=ilnik@webrtc.org,sprang@webrtc.org,kron@webrtc.org
Change-Id: I8960a6cc15e552925129ba0037f197ff3fd93c25
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134100
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27737}
The function iterated over two containers, destroyed their elements
and popped those elements one at a time. It's more efficient to
destroy all of the elements, then clear() the container.
Bug: None
Change-Id: I17aa88694ee41df64c5793b08b96899b7ff04071
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133901
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27730}
Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.
Also, added some missing tests.
Bug: webrtc:10460
Change-Id: Ia49072c3041e75433f125a61050d2982b2bec1da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27719}
Rename "UpdateLayerConfig" to the more appropriate "NextFrameConfig".
Also update some comments in vp8_frame_buffer_controller.h.
Bug: None
Change-Id: Iba8227f84e33e5ebd28d2eeb10fe03e776036603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133202
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27660}
A typo in a previous CL made OnLossNotification() accept its
single argument as a const-value, rather than a const-reference.
Bug: webrtc:10501
Change-Id: I5e6f9c79f15205b75ec90a53d3fccf3dd9927e33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133343
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27659}
Prior to this CL, this was indicated by passing |size_bytes| = 0
to the method.
Bug: webrtc:10501
Change-Id: Icff3bb83344834dc62d62bde5ec5d05096a08e11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132712
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27620}
This allows picking up the output in Android tests, where stdout/stderr
is lost but RTC_LOGs are picked up by the org.webrtc.Logging utility.
Tested: Downstream Android tests.
Bug: webrtc:10349
Change-Id: I1379f4303640dbc9621c64d9c88cf61bc8447ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132704
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27616}
This CL adds an experiment where aggressiveness of the rate controller
is tuned based on if the application is network constrained or not.
Bug: webrtc:10155
Change-Id: I6c8cd116f57321c5b36cf5a69840913936091aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132786
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27615}
The former became redundant and didn't guarantee
numerical stability for variance computation.
Bug: webrtc:10412
Change-Id: Idc291abe9add41bde9e7734f179e5d6c65f2630b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132460
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27605}
This is a reland of 7ac0d5f348
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
profile-level-id for H.264 comes in through the SdpVideoFormat,
rather than through these members.
Bug: None
Change-Id: I9c4ea8873346ca16174aecf5f90a649cbaf913dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132545
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27571}
This reverts commit 7ac0d5f348.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
After https://webrtc-review.googlesource.com/c/src/+/131141 there are some minor
changes to the encoding performance, hence the updated values.
Bug: none
Change-Id: Ifa661eea15a0d52f4760f4aac9294074faab757f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27463}
This fixes a regression introduces way back in August 2018:
https://webrtc-review.googlesource.com/c/src/+/91863/
For bonus points, also fixing an auxiliary test issue.
Bug: webrtc:10479, webrtc:10260
Change-Id: I4e99fe6e070446d10357d9d1a9d1ffc9dedcf419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129926
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27409}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
In this CL:
- Created static helper function GenericFrameInfo::DecodeTargetInfo to
convert DTI symbols to a list of GenericFrameInfo::OperatingPointIndication.
- Added per frame DTI information for the different stream structures.
Bug: webrtc:10342
Change-Id: I62ff2e9fc9b380fe1d0447ff071e86b6b35ab249
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129923
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27350}
- Add GetFrameStatistics API:
This is useful for downstream test users that want to read frame-level stats.
- Remove other APIs that are not used by downstream tests:
* AddFrame
* GetFrame
* GetFrameWithTimestamp
* SliceAndCalcAggregatedVideoStatistic
* PrintFrameStatistics
* Size
* Clear
The implementations, which are used by the fixture implementation, are kept.
Bug: webrtc:10349
Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27306}
