This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.
It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.
Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
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.
(This is a reland without changes after updates to downstream projects)
Original-Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153483
Bug: webrtc:10977
Change-Id: I02459c5982da2e0542a837514f5753c5f96401c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154355
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29330}
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}
Injection is made possible through VP8Encoder::Create.
According to native-api.md, it is a defacto public API despite
not being in the api/ folder.
Bug: webrtc:10259, webrtc:10382
Change-Id: Ifc5d55aa99613cfee0fcb4f0c6690121c85b2e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27281}
This allows offline visualization of the different TL.
For now, there is no need to do the same for the decoded frames.
Bug: webrtc:10349
Tested: 1) ninja -C out/Debug; and out/Debug/modules_tests --gtest_filter="*MultiresVP8*:*SvcVP9*". 2) Downstream tests.
Change-Id: Iaf5ab19ee681488706d8777a5adba78efd5cc1ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128861
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27240}
This CL paves the way to making FrameBufferController injectable.
LibvpxVp8Encoder can manage multiple streams. Prior to this CL,
each stream had its own frame buffer controller, all of them held
in a vector by LibvpxVp8Encoder. This complicated the code and
produced some code duplication (cf. SetupTemporalLayers).
This CL:
1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs
will make this factory injectable.)
2. Makes LibvpxVp8Encoder use a single controller. This single
controller will, in the case of multiple streams, delegate
its work to multiple controllers, but that fact is not visible
to LibvpxVp8Encoder.
This CL also squashes CL #126046 (Send notifications of RTT and
PLR changes to Vp8FrameBufferController) into it.
Bug: webrtc:10382
Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27206}
To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor
TaskQueueForTest can be used instead in tests and keep the convenient constructor.
Also cleanup the TaskQueueForTest a bit:
move the class to webrtc namespace
add default constructor
disallow copy using language construct instead of macro
cleanup build dependencies
rename build target (to match move out of the rtc namespace)
Bug: webrtc:10284
Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27193}
This is a reland after changes to the downstream project
VP9 screenshare is not used currently, and with these values according
to local testing with screenshare_loopback, we get performance not worse than current vp8 settings for similar uplink and downlink values.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
Bug: webrtc:10257
Change-Id: Ib21d7678bd839a3c47457515b0d768c0b979ea40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126524
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27040}
This reverts commit 12abf671fd.
Reason for revert: Breaks downstream project.
Original change's description:
> Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"
>
> This is a reland without any changes as it seems problems with webrtc-in-chrome importer were flakes or
> caused by some issues within chrome codebase.
>
> Tune vp9 screenshare bitrate and framerate of spatial layers
>
> VP9 screenshare is not used currently, and with these values according
> to local testing with screenshare_loopback, we get performance not worse than current vp8 settings for similar uplink and downlink values.
>
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
>
> Bug: webrtc:10257
> Change-Id: Ie819d8bbab4f14877daac733d162e5ae7ebf2a8e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126460
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27036}
TBR=ilnik@webrtc.org,jeroendb@webrtc.org,kron@webrtc.org
Change-Id: I9ad9017b054213f931b3b39c641060d35565f17d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126523
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27037}
This is a reland without any changes as it seems problems with webrtc-in-chrome importer were flakes or
caused by some issues within chrome codebase.
Tune vp9 screenshare bitrate and framerate of spatial layers
VP9 screenshare is not used currently, and with these values according
to local testing with screenshare_loopback, we get performance not worse than current vp8 settings for similar uplink and downlink values.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
Bug: webrtc:10257
Change-Id: Ie819d8bbab4f14877daac733d162e5ae7ebf2a8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126460
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27036}
This reverts commit aaf3cb3adb.
Reason for revert: Chrome importer consitently failing after this change
Original change's description:
> Tune vp9 screenshare bitrate and framerate of spatial layers
>
> VP9 screenshare is not used currently, and with these values according
> to local testing with screenshare_loopback, we get performance not worse
> than current vp8 settings for similar uplink and downlink values.
>
> Bug: webrtc:10257
> Change-Id: Icabac04fbd3d616412bbae59291a1fc026d0a504
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27023}
TBR=ilnik@webrtc.org,kron@webrtc.org
Change-Id: I1ef1eeec8fe87a7662a354ef6362b7d463b2bb4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126340
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27027}
VP9 screenshare is not used currently, and with these values according
to local testing with screenshare_loopback, we get performance not worse
than current vp8 settings for similar uplink and downlink values.
Bug: webrtc:10257
Change-Id: Icabac04fbd3d616412bbae59291a1fc026d0a504
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126226
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27023}
These are used by the test runner to pick up perf values
to be shown in the perf dashboard.
Bug: webrtc:10349
Change-Id: Ib3b2479f7a20b66192751bee8237d757f5870bd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126220
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27020}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
Vp8FrameBufferController is currently just a renamed Vp8TemporalLayers,
but subsequent CLs will modify Vp8FrameBufferController in ways that are
not relevant for Vp8TemporalLayers. Namely:
1. Loss notifications will be added.
2. Packet-loss rate will be tracked.
3. RTT will be tracked.
4. Vp8FrameBufferController will be made injectable.
Vp8TemporalLayers is retained in order to:
1. Avoid needlessly changing api/.
2. Place for code shared between DefaultTemporalLayers and ScreenshareLayers.
We can remove it in the future (with a proper public announcement).
Bug: webrtc:10382
Change-Id: I49ad1b9bc1954d51bb0b5e60361985f1eb12ae9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126045
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27009}
If minQP is reached and encoder undershoot consistently, we consider the
quality good enough and throttle encode frame rate.
Bug: webrtc:10310
Change-Id: Ifd07280040dd67ef6e544efdd4619d47bff951e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125461
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27003}
If minQP is reached and encoder undershoot consistently, we consider the
quality good enough and throttle encode frame rate.
This CL also adds perf tests for high fps vp9 screenshare.
Bug: webrtc:10310
Change-Id: I49fc7d31f9f596a9ecb5f85fe9e0c7861d4915f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125761
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26997}
In this CL:
- Updated Vp8TemporalLayers::OnEncodeDone to take a CodecSpecificInfo
instead of a CodecSpecificInfoVP8, so that both the VP8 specific and
generic information can be populated.
- Added structs to represent the GFD template structure.
- Added code to generate templates for video/screensharing.
Bug: webrtc:10342
Change-Id: I978f9d708597a6f86bbdc494e62acf7a7b400db3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123422
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26987}
For a single layer vp9, the target bitrate was not set correctly. This
may cause a problem for screenshare case, since target bitrate is
respected in that case. If it were less than a min bitrate, the only
spatial layer was permanently disabled.
Bug: webrtc:10257
Change-Id: I0980349adfc2970f810acc51a3e2a31ecbb2bbd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125681
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26970}
there's no easy way to inject the Clock in ScreenshareLayers under
normal use. To allow faking the clock, rtc::TimeMillis is used instead.
Bug: webrtc:10365
Change-Id: I46c7f76514672190a0f0f5816a2c858bc6c76fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/125189
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26946}
CodecSpecificInfo has a default constructor, so initializing by memset is not necessary and is in the way of adding non-trivial members.
Related chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/1495533
Bug: webrtc:10342
Change-Id: I36046f919f5fc34ea51de7288ff5c9cc0f2950b8
Reviewed-on: https://webrtc-review.googlesource.com/c/125093
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26924}
Simulcast screenshare appears broken due to unrelated changes. It
implicitly relied on SimulcastEncoderAdapter fallback, which happened before
if streams had same resolution. It's not the case anymore. Thus, this CL
adds checks for different frame-rate in simulcast streams.
FullStackTests are also updated to use actual parameters.
Bug: none
Change-Id: I2c1ddb1b39edb96464a0915dfcb9cb4e18844187
Reviewed-on: https://webrtc-review.googlesource.com/c/124494
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26869}
This quality boost means that we sometimes drop a _lot_ of frames in the
base layer. It also interacts poorly with the bitrate adjuster since
even if frames are dropped they are often over-sized.
The setting still leaves the current behavior as default, but can be
changed using the WebRTC-VideoRateControl field trial.
Bug: webrtc:10155
Change-Id: I1a92ec69bab61b5148fe9d8bc391ac5ee1019367
Reviewed-on: https://webrtc-review.googlesource.com/c/122840
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26659}
This CL continues the work began by CL #119958, extending it
to ScreenshareLayers.
Bug: webrtc:10249
Change-Id: I59d0c062a93b288007977e00aa3a2e0929509e0c
Reviewed-on: https://webrtc-review.googlesource.com/c/120042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26526}
Prior to this CL, when software VP8 encoding was done with one temporal
layer, instead of only predicting from the latest frame, the code
allowed the encoder to reference the latest key frame as well.
This improves quality for the few frames immediately after
the key frame, but is not useful for later frames, which diverge
significantly from the key frame. However, the cost of producing
the prediction from more than one reference is incurred by all frames.
My measurements of the effect of this show an improvement
in CPU utilization of 5%-13% when this is not done.
foreman_352x288, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(566.187, 570.012, 575.665) = 570.621
send_avg_qp: 45.36
send_avg_psnr: 37.13
Post-change:
send_enc_speed_fps: avg(633.188, 604.694, 623.232) = 620.371
send_avg_qp: 45.88
send_avg_psnr: 37.0749
Improvement in send_enc_speed_fps: 8.71%
foreman_480x272, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(481.244, 486.971, 487.322) = 485.179
send_avg_qp: 48.9
send_avg_psnr: 37.6217
Post-change:
send_enc_speed_fps: avg(521.651, 499.416, 511.551) = 510.872
send_avg_qp: 48.88
send_avg_psnr: 37.6094
Improvement in send_enc_speed_fps: 5.29%
news_352x288, 30fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(699.407, 697.837, 699.49) = 698.9113333
send_avg_qp: 24.15
send_avg_psnr: 40.9551
Post-change:
send_enc_speed_fps: avg(758.526, 768.104, 757.232) = 761.2873333
send_avg_qp: 23.9833
send_avg_psnr: 40.9697
Improvement in send_enc_speed_fps: 8.92%
Bridge_180x320_15 (video of brandtr@ from Google), 15fps, target bitrate 500kps
Pre-change:
send_enc_speed_fps: avg(454.757, 450.399, 446.812) = 450.656
send_avg_qp: 17.6771
send_avg_psnr: 39.9267
Post-change:
send_enc_speed_fps: avg(500.014, 513.316, 513.613) = 508.981
send_avg_qp: 17.6837
send_avg_psnr: 39.9137
Improvement in send_enc_speed_fps: 12.94%
Bug: webrtc:10281
Change-Id: If02736e1535c5f46689fd42b657e35a1e1f64d6d
Reviewed-on: https://webrtc-review.googlesource.com/c/120904
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26511}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
FrameConfig is not specific to temporal layers. Anything that
can control referenced/updated buffers could potentially use it.
Bug: webrtc:10259
Change-Id: I04ed177ee884693798c3b69e35fd4255ce1e9062
Reviewed-on: https://webrtc-review.googlesource.com/c/120355
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26448}
Prior to this CL, RtpPayloadParams had code that assumed
dependency patterns in VP8, in order to write that information
into the [Generic Frame Descriptor] RTP extension.
This CL starts moving that code out of RtpPayloadParams.
Upcoming CLs will migrate additional encoder-wrappers to
the new scheme, then remove the deprecated code.
Bug: webrtc:10249
Change-Id: I5fc84aedf8e11f79d52b989ff8b7ce9568b6cf32
Reviewed-on: https://webrtc-review.googlesource.com/c/119958
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26438}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
This flag only needs to be set in kOn interlayer prediction mode, because
in all others, if new layer is enabled - a keyframe is generated.
Also, use external reference control in that case, because libvpx creates
rtp-incompatible references in that case.
Bug: webrtc:10180
Change-Id: I0fad188fa8cd424f831bac219769dbad3a788b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/118041
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26316}
This CL add new data to the VideoEncoder::EncoderInfo struct, indicating
how the encoder intends to allocate frames across spatial and temporal
layers.
This metadata will be used in upcoming CLs to control how the encoder's
rate controller performs.
Bug: webrtc:10155
Change-Id: Id56fae04bae5f230d1a985171097d7ca83a3be8a
Reviewed-on: https://webrtc-review.googlesource.com/c/117900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26300}
Use size() accessor function. Also replace most nearby uses of _buffer
with data().
Bug: webrtc:9378
Change-Id: I1ac3459612f7c6151bd057d05448da1c4e1c6e3d
Reviewed-on: https://webrtc-review.googlesource.com/c/116783
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26273}
Currently, CPU overuse settings for HW encoders are sometimes being used
even though the actual encoder is a SW encoder, e.g. in case of SW fallback
when the encoder is initialized. Polling is_hardware_accelerated after the
encoder has been created and initialized will improve choosing the correct
CPU overuse settings.
Bug: webrtc:10065
Change-Id: Ic6bd67630a040b5a121c13fa63dd074006973929
Reviewed-on: https://webrtc-review.googlesource.com/c/116688
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26266}
Previous attempt: https://codereview.webrtc.org/1882733006/. There
might be some benefit of having dummy encoder/decoder available in
video_loopback.
Bug: webrtc:5791
Change-Id: Iec316296754178c92b18dd3cf92f67ce6aed9439
Reviewed-on: https://webrtc-review.googlesource.com/c/112596
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26043}
Without the added preprocessor check, iOS device will be using the desktop logic to determine the number of thread. This put iPhone 8 and iPhone X to use 3 threads and all other iPhones after iPhone 5 to use a single thread.
This CL added a preprocessor for WEBRTC_IOS to have it own thread number calculation logic. In which, the maximum number of thread is fetched from a field_trial and capped by the number of CPU available on the device.
Bug: webrtc:10005
Change-Id: I8c6257fcbf85b07bc986b5f733dbabb3feee37f7
Reviewed-on: https://webrtc-review.googlesource.com/c/110941
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25997}
This is a reland of 4c0cc5bc5f
I added more Chrome checks for munging profiles in the below patch
that will allow us to land this without regressions.
https://chromium-review.googlesource.com/c/chromium/src/+/1366898
Original change's description:
> Reland Profile 2 to default profiles
>
> This is a reland after chrome browser tests are updated.
>
> Bug: webrtc:9376
> Change-Id: I818bf5d447da7901ffe49f2c452decb89196e829
> TBR: niklas.enbom@webrtc.org
> Reviewed-on: https://webrtc-review.googlesource.com/c/112060
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25778}
Bug: webrtc:9376
Change-Id: I8998537816a773961e519535c6afdde3801b5918
TBR: niklas.enbom@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/113980
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25977}
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.
* Move calculation of padding bitrate to SvcRateAllocator class.
* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.
Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
Read using capacity() method, write using set_buffer() method. This is
a preparation for making the member private, and renaming it to
capacity_.
Bug: webrtc:9378
Change-Id: I2f96679d052a83fe81be40301bd9863c87074640
Reviewed-on: https://webrtc-review.googlesource.com/c/113520
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25934}
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
and return several frames combined from FrameBuffer.
Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
See https://ci.chromium.org/p/chromium/builders/luci.chromium.webrtc/WebRTC%20Chromium%20Mac%20Tester
First, we figured that "ba2840c Various VP9 high fps fixes by Ilya Nikolaevskiy" was the cause and it was reverted but it did not help.
We must now try the other CL which had done changed in VP9.
Revert "Reland Profile 2 to default profiles"
This reverts commit 4c0cc5bc5f.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Reland Profile 2 to default profiles
>
> This is a reland after chrome browser tests are updated.
>
> Bug: webrtc:9376
> Change-Id: I818bf5d447da7901ffe49f2c452decb89196e829
> TBR: niklas.enbom@webrtc.org
> Reviewed-on: https://webrtc-review.googlesource.com/c/112060
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25778}
TBR=emircan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9376
Change-Id: I3eb935c08341ce51fa16717ed7b3be5f5253aa2f
Reviewed-on: https://webrtc-review.googlesource.com/c/112597
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25874}
kInvalid does not have a corresponding entry in the standard is therefore removed.
kUNSPECIFIED should be used instead.
Bug: webrtc:8651
Change-Id: Iee8cd85830aedaa4a9102251121b9975d40fa5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/112421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25871}
This is the first step in moving the metadata and eventually replacing
VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo.
Bug: webrtc:10065
Change-Id: If925b895718e1b1225d2cf49bede1adb3ff281b8
Reviewed-on: https://webrtc-review.googlesource.com/c/112285
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25856}
This reverts commit ba2840ce4e.
Reason for revert: Looks like this breaks all VP9 tests on the Chromium level, for Mac: https://ci.chromium.org/buildbot/chromium.webrtc/Mac%20Tester/85866
Search for TIMED OUT in for instance https://logs.chromium.org/logs/chromium/bb/chromium.webrtc/Mac_Tester/85866/+/recipes/steps/browser_tests/0/stdout (it times out because the video is frozen).
Original change's description:
> Various VP9 high fps fixes
>
> - Enable flexible mode in loopback tools and quality tests
> - Ensure duplicate references are not set by the sender in video header
> - Reset first active spatial layer on keyframe in encoder
> - Make vp9 encoder to not generate spatial references for first active
> layer with external reference control in svc flexible mode
>
> Bug: webrtc:10049
> Change-Id: If9ff576ea8a1a2fef6116b17b5b5adff08c5f8c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/112080
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25795}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10049
Change-Id: Ie6a7daf6414337173fec38c5ff546d509951cba6
Reviewed-on: https://webrtc-review.googlesource.com/c/112400
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25842}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
- Enable flexible mode in loopback tools and quality tests
- Ensure duplicate references are not set by the sender in video header
- Reset first active spatial layer on keyframe in encoder
- Make vp9 encoder to not generate spatial references for first active
layer with external reference control in svc flexible mode
Bug: webrtc:10049
Change-Id: If9ff576ea8a1a2fef6116b17b5b5adff08c5f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/112080
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25795}
Pass encoded frames to packetizer immediately if encoder is configured
to drop whole superframe.
Bug: webrtc:9950
Change-Id: Iedee9618bb146307efd5a86cb35bf14b5e64b341
Reviewed-on: https://webrtc-review.googlesource.com/c/109002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25771}
Move HdrMetadata to ColorSpace as part of preparing for joint transmission
of these two objects.
Bug: webrtc:8651
Change-Id: Ie948011a2c0106d5967cb5ef3b9565217e798272
Reviewed-on: https://webrtc-review.googlesource.com/c/111481
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25730}
This increases expected value of maximum buffer level in VP8/9 tests
up to 1 second and thus alignes it with the value that WebRTC uses by
default for these codecs.
Bug: webrtc:10017
Change-Id: I8fd41e8006f11c230d844a053c04656408c2ec97
Reviewed-on: https://webrtc-review.googlesource.com/c/111503
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25716}
Libvpx has been recently updated and this test was failing because
of a slightly different value.
TBR=sprang@webrtc.org
Bug: webrtc:10017
Change-Id: I5fe9161eef5c3e1ff8e0dceb36a663648d8f4617
Reviewed-on: https://webrtc-review.googlesource.com/c/111461
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25709}
Instead of optionally compile VP9 source files based on the value of
the GN argument 'rtc_libvpx_build_vp9', this CL uses the preprocessor
macro RTC_ENABLE_VP9 to decide if VP9 related code needs to be compiled
or not.
Bug: None
Change-Id: I5c1b69d7ec35e8446181d98c912277d0ae8fdba2
Reviewed-on: https://webrtc-review.googlesource.com/c/111063
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25685}
RTC_ENABLE_VP9 is more natural to deal with then RTC_DISABLE_VP9.
In all the places this macro is used, WebRTC needs to do more things
so it is easier to "do more if RTC_ENABLE_VP9 is defined" than
"do more if RTC_DISABLE_VP9 is not defined".
Bug: None
Change-Id: If992e5c554173e6af3f030f6e0fd21bd82acf9eb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/111242
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25679}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
If rate controller is trusted, we disable the frame dropper in the
media optimization module.
This is a re-land of
https://webrtc-review.googlesource.com/c/src/+/105020
Bug: webrtc:9890
Change-Id: I418e47a43a1a98cb2fd5295c03360b954f2288f2
Reviewed-on: https://webrtc-review.googlesource.com/c/109141
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25570}
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
This is a cleanup CL related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: I3d879196af83856ece1418fa786aab03a3dd3c8c
Reviewed-on: https://webrtc-review.googlesource.com/c/108820
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25466}
This is just a cleanup CL related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: I9a8347aa382bf44f3cd6c38d89bea0e9d68a50e0
Reviewed-on: https://webrtc-review.googlesource.com/c/108781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25464}
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705
Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
Bug: webrtc:9419
Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94
Reviewed-on: https://webrtc-review.googlesource.com/c/106343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25207}
Only used for output filename nowadays. Previously, it was used for
selecting the codec implementation. That is now done by injecting
the appropriate codec factory.
Bug: webrtc:9317
Change-Id: Ia2bf28f7df165fb65410ecd1f5d646ee6604e1be
Reviewed-on: https://webrtc-review.googlesource.com/c/106023
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25204}
Move MockVideoDecoder from
modules/video_coding/include/mock/mock_video_codec_interface.h
to
api/test/mock_video_decoder.h
The mock encoder has already moved:
https://webrtc-review.googlesource.com/c/src/+/105620
Keeping the old header until downstream projects have been updated.
Bug: webrtc:9722
Change-Id: I4bc849173a04813064212f17761876695ca3fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/105900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25170}
Also renaming it Vp8TemporalLayers to show that it is codec specific.
Bug: webrtc:9012
Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f
Reviewed-on: https://webrtc-review.googlesource.com/c/104643
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25137}
If rate controller is trusted, we disable the frame dropper in the
media optimization module.
Bug: webrtc:9722
Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/105020
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25107}
I don't think this has any impact, just wanted to have a first unit
test to play around with.
Bug: None
Change-Id: I892e2642f0243c5c9ee807cf71abcd96112b25f4
Reviewed-on: https://webrtc-review.googlesource.com/c/105000
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25089}
Set number of decode threads equal to number of available cores and
limit the maximum value to the maximum number of tiles possible for
HD resolution.
Bug: webrtc:9829, b/117291409
Change-Id: Ib5ccd5cc412011d4438258491efc060cdd050fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/104064
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25059}
- Added field trial to force issuing of key frame on deactivation of
spatial layer. This fixes video corruptions in VP9 K-SVC tests where
layers can be activated/deactivated on-fly due to bandwidth change.
- Added 100ms network delay to the test with restricted link capacity.
This fixes rapid drop of available bandwidth which happens when
bandwidth overuse is detected in the very beginning of call and several
feedback packets arrive without any delay. Also, this makes the test
more realistic.
- Disabled filtering of spatial layer in the test with restricted
link capacity. 1) We don't really need filtering in this test.
2) It appeared that in video quality tests filtering is done before
sending packets to network simulator. Filtering of high layers causes
channel underuse which is compensated by increase of sent bitrate.
This is why we got sent/media bitrates about 2Mbps in test where link
was limited to 1Mbps.
Bug: chromium:889017
Change-Id: I33ffcee0274523f6183c3bbd27d3d29395417d52
Reviewed-on: https://webrtc-review.googlesource.com/c/103520
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24988}
This refactoring merged PopulateCodecSpecific and FrameEncoded into a
single callback method. It also removes the FrameConfig parameter and
instead relies on the temporal layer to remember it internally.
Bug: webrtc:9012
Change-Id: I489b76821b534398ad452643f1322f411d3455b1
Reviewed-on: https://webrtc-review.googlesource.com/95681
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24957}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
Mark all low spatial layer frames as references (not just frames of
active low spatial layers) if inter-layer prediction is enabled since
these frames are indirect references of high spatial layer, which can
later be enabled without key frame.
Bug: webrtc:9782
Change-Id: Iffa5039fab2673a5582e7cdc9be4a36d9e8deb63
Reviewed-on: https://webrtc-review.googlesource.com/102063
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24849}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
VP9 frame rate controller is supposed to be used in screen mode only
but it was partially enabled in normal video mode. This restricts use
of VP9 frame rate controller to screen mode.
Bug: chromium:884164
Change-Id: Ie2eaa31f3364a8abccbc4171007708cf7040fc38
Reviewed-on: https://webrtc-review.googlesource.com/100424
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24769}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
Set target frame rate of spatial layer equal to minimum of two: maximum
frame rate of layer (SpatialLayer::maxFramerate) and maximum frame rate
of codec (VideoCodec::maxFramerate).
Bug: webrtc:9740, webrtc:9739, chromium:882358
Change-Id: I34f36e7fd2889f0417474347abab5327fa2d9d7c
Reviewed-on: https://webrtc-review.googlesource.com/99501
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24686}
This is a reland of ae9e188e67
Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}
TBR=sprang@webrtc.org
Bug: webrtc:9682
Change-Id: Idcce315890c79301da532f9ba4997e9606f73fb0
Reviewed-on: https://webrtc-review.googlesource.com/99340
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24669}
This reverts commit ae9e188e67.
Reason for revert: Verify if this causes chromium:882358.
Original change's description:
> Frame rate controller per spatial layer.
>
> This allows VP9 encoder wrapper to control frame rate of each spatial
> layer. The wrapper configures encoder to skip encoding spatial layer
> when actual frame rate exceeds the target frame rate of that layer.
> Target frame rate of high spatial layer is expected to be equal or
> higher then that of low spatial layer. For now frame rate controller
> is only enabled in screen sharing mode.
>
> Added unit test which configures encoder to produce 3 spatial layers
> with frame rates 10, 20 and 30fps and verifies that absolute delta of
> final and target rate doesn't exceed 10%.
>
> Bug: webrtc:9682
> Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
> Reviewed-on: https://webrtc-review.googlesource.com/96640
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24593}
TBR=sprang@webrtc.org,ssilkin@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9682, chromium:882358
Change-Id: Idc4051eef72104823038ed9139bb9c75018f7d86
Reviewed-on: https://webrtc-review.googlesource.com/99082
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24646}
Today, the internal frame dropper in libvpx vp8 encoder is enabled or
disabled based on video or screen content. This is then expected to
match up with screenshare vs default temporal layers implementation.
This cl makes libvpx query the temporal layers implementation as well,
breaking this implicit dependency and allows frames to be dropped if
default temporal layers is used with screen content.
Bug: webrtc:9734
Change-Id: If2523a211f4929f16e65a02fa7a6b4edf7328571
Reviewed-on: https://webrtc-review.googlesource.com/99062
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24645}
This CL performs some cleanups on multiplex files:
- Adds more comments to factory about usage.
- Moves image packer outside /include as it doesn't need to be public.
- Other small lint issues.
Bug: webrtc:9632
Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32
Reviewed-on: https://webrtc-review.googlesource.com/98700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24615}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This allows VP9 encoder wrapper to control frame rate of each spatial
layer. The wrapper configures encoder to skip encoding spatial layer
when actual frame rate exceeds the target frame rate of that layer.
Target frame rate of high spatial layer is expected to be equal or
higher then that of low spatial layer. For now frame rate controller
is only enabled in screen sharing mode.
Added unit test which configures encoder to produce 3 spatial layers
with frame rates 10, 20 and 30fps and verifies that absolute delta of
final and target rate doesn't exceed 10%.
Bug: webrtc:9682
Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
Reviewed-on: https://webrtc-review.googlesource.com/96640
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24593}
This feature went to stable with M69. Switch is in M69 and M70 banches.
Since tot is now M71 and we have not seen any issues, let's clean this
up.
Bug: webrtc:9634
Change-Id: I708bab55b0443d0873b09dd5b71cdfad72397a7a
Reviewed-on: https://webrtc-review.googlesource.com/98002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24581}
This is a reland of da0898dfae
Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}
Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
Replaces the VP8 simulcast index and VP9 spatial index formely part of
CodecSpecificInfo.
Bug: webrtc:9378
Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
Reviewed-on: https://webrtc-review.googlesource.com/83161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24485}
It's reasonable to allow clients implementing their own VideoCodecTests
to decide wether they should run in real-time.
Removes the IsAsyncCodec helper, as the assumptions it made are outdated,
and it is no longer useful.
Bug: None
Change-Id: If766935d4947555af54f499a30cfe553bde4c1ab
Reviewed-on: https://webrtc-review.googlesource.com/95722
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24478}
This will allow us to configure VP9 encoder to produce spatial layers
with different frame rates.
Bug: webrtc:9650
Change-Id: I3a9c58072003b8a8da681d5291d8f7ede7f52fa4
Reviewed-on: https://webrtc-review.googlesource.com/95427
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24435}
- Allow use of flexible mode which was blocked in webrtc:9261 since it
only worked together with old screen sharing. Since webrtc:9244 flexible mode
works with both normal and screen coding modes.
- Add unit test that checks that reference list encoder writes into RTP
payload descriptor and the predefined one match.
Bug: webrtc:9585
Change-Id: I4a1bdc51cbf15e7224cc7c271af8b2e3d46657d1
Reviewed-on: https://webrtc-review.googlesource.com/94778
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24355}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
It didn't account for implicit bitrate allocation, which is used in
some unit tests, when bitrate distribution is done by the encoder
wrapper.
Bug: none
Change-Id: I8fcf28e10f7a6c378580ef917221ad5c8d3869c9
Reviewed-on: https://webrtc-review.googlesource.com/94775
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24343}
This CL is a minimum effort/low risk fix.
Later CLs take a more thorough approach.
Bug: webrtc:9634
Change-Id: I728a061a4e71b38a559ee438646de83cd2cb3517
Reviewed-on: https://webrtc-review.googlesource.com/94760
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24326}
kAllBuffers in default_temporal_layers.cc introduces a static initializer,
that is banned in Chromium, and that blocks WebRTC roll into Chromium.
This CL removes it to unblock.
Bug: webrtc:9012
Change-Id: Ide181f63d85748dc2d09199024f1b80868d485fd
Reviewed-on: https://webrtc-review.googlesource.com/94460
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24307}
Multiplex encoder is now supporting attaching user-defined data to the video
frame. This data will be sent with the video frame and thus is guaranteed to
be synchronized. This is useful in cases where the data and video frame need
to by synchronized such as sending information about 3D objects or camera
tracking information with the video stream
Multiplex Encoder with data is implemented in a modular way. A new
VideoFrameBuffer type is created with the encoder. AugmentedVideoFrameBuffer
holds the video frame and the data. MultiplexVideoEncoder encodes both
the frame and data.
Change-Id: I23263f70d111f6f1783c070edec70bd11ebb9868
Bug: webrtc:9632
Reviewed-on: https://webrtc-review.googlesource.com/92642
Commit-Queue: Tarek Hefny <tarekh@google.com>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24297}
This CL introduces a few changes to the default VP8 temporal layers:
* The pattern is now reset on keyframes
* The sync flag is inferred rather than hard-coded
* Support is added for buffer search order
Bug: webrtc:9012
Change-Id: Ice19d32413d20982368a01a7d2540d155e185ad4
Reviewed-on: https://webrtc-review.googlesource.com/91863
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24288}
* Make shorter 4-frame pattern default if 2 temporal layers are used.
* Make DefaultTemporalLayers usable by upper simulcast stream with 2tl.
* If experimental settings are enable, bump the max bitrate for the top
stream. Since we're now using probing everywhere the rampup should be
less of an issue.
* Additionally, fixes an issue in full stack tests, where
ScopedFieldTrials in an experiment would override the
--force_fieldtrials specified at command line. Some trials added by
the test bots caused timeouts without this.
Bug: webrtc:9477
Change-Id: I42410605d416b51c4fbfe5b6b850997484af583c
Reviewed-on: https://webrtc-review.googlesource.com/92883
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24252}
This reverts commit fad2aa23b4.
Reason for revert: There seems to be a mismatch with Chrome's default for VP8.
Original change's description:
> Extract color space from Vp8 decoder
>
> Makes use of ColorSpace class to extract info from Vp8 stream.
>
> Bug: webrtc:9522
> Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
> Reviewed-on: https://webrtc-review.googlesource.com/90183
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24086}
TBR=sprang@webrtc.org,emircan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9522
Change-Id: Ie589963159c9e7ccbc52bf3fdfcbc383656a4ca9
Reviewed-on: https://webrtc-review.googlesource.com/92500
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24191}
It was moved to api/video_codecs/video_encoder_config.h in cl
https://webrtc-review.googlesource.com/77683.
Bug: webrtc:8830
Change-Id: I197fd3270d3dea0a5ec98b22cc675c407c388e93
Reviewed-on: https://webrtc-review.googlesource.com/90243
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24092}
Makes use of ColorSpace class to extract info from Vp8 stream.
Bug: webrtc:9522
Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
Reviewed-on: https://webrtc-review.googlesource.com/90183
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24086}
Makes use of ColorSpace class to extract info from H264 stream.
Bug: webrtc:9522
Change-Id: I651d16707260bb2867b1eda95dd4956d62c47279
Reviewed-on: https://webrtc-review.googlesource.com/90180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24085}
In order to have public video bitrate allocator factory, the video bitrate allocator has be part of
the api.
Bug: webrtc:9513
Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2
Reviewed-on: https://webrtc-review.googlesource.com/88083
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#24073}
This CL is the first step for introducing color space information in webrtc.
- Add ColorSpace class listing color profiles.
- Add ColorSpace as a member of webrtc::VideoFrame.
- Make use of this class by extracting info from VP9 decoder.
Bug: webrtc:9522
Change-Id: I5e2514efee2a193bddb4459261387f2d40e936ad
Reviewed-on: https://webrtc-review.googlesource.com/88540
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23988}
Value 10 seems to be too small for some implementations. Updates the
value to 20. This affects VideoCodecTestFixture.
Bug: None
Change-Id: Ibbeb7cb5ef23f8ac625d37aaa764c9d245f23e9d
Reviewed-on: https://webrtc-review.googlesource.com/87562
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23882}
This is a step toward simplifying the VideoCodec struct and removing the
targetBitrate. The hard-coded values now reside in
SimulcastRateAllocator.
A follow-up will do away with the field altogether.
Bug: webrtc:9504
Change-Id: I74d483682309d363048fbbbd31e0607d7242f504
Reviewed-on: https://webrtc-review.googlesource.com/87424
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23876}
This reverts commit 425193b4a9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Unit test for case where the number of active and configured spatial"
>
> This reverts commit 5eb6045ce5.
>
> Reason for revert: Test breaks downstream.
>
> Original change's description:
> > Unit test for case where the number of active and configured spatial
> > layers differ.
> >
> > Bug: webrtc:9472
> > Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> > Reviewed-on: https://webrtc-review.googlesource.com/85644
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23782}
>
> TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
>
> Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9472
> Reviewed-on: https://webrtc-review.googlesource.com/86320
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23785}
TBR=brandtr@webrtc.org,terelius@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9472
Change-Id: I796909c553702a0fa19e5e16e4586f915569b134
Reviewed-on: https://webrtc-review.googlesource.com/87220
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23847}
Since the frame is processed on the same thread as the decoding happens
on, keeping a reference to the frame may cause deadlocks on some
implementations.
Longer term, we should probably move the frame processing to a separate
thread but this is an easy fix for now.
Bug: b/110246814
Change-Id: I251737e2188e1755d45b35165586d1b0daf14595
Reviewed-on: https://webrtc-review.googlesource.com/87104
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23843}
Also adjust to base-layer fraction for the shortened 3-tl pattern to be
60%, just like the 2-tl setting.
This CL removes direct use of the allocation matrix and moves it behind
a static getter.
Bug: webrtc:9477
Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5
Reviewed-on: https://webrtc-review.googlesource.com/86681
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23834}
This CL adds VP9 profile information in SDP. It adds the necessary fields and
enums to codec containers.
Additional profiles will be followed.
Bug: webrtc:9376
Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54
Reviewed-on: https://webrtc-review.googlesource.com/81960
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23810}
This reverts commit 5eb6045ce5.
Reason for revert: Test breaks downstream.
Original change's description:
> Unit test for case where the number of active and configured spatial
> layers differ.
>
> Bug: webrtc:9472
> Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> Reviewed-on: https://webrtc-review.googlesource.com/85644
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23782}
TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org
Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9472
Reviewed-on: https://webrtc-review.googlesource.com/86320
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23785}
The variable, num_active_spatial_layers, is used to construct ssData.
This CL reverts two instances of num_active_spatial_layers not
related to ssData construction.
Bug: None
Change-Id: I4d90d4578684dfdf8bd5a39c7a2fe778fce4414c
Reviewed-on: https://webrtc-review.googlesource.com/85643
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23756}
This removes call of av_register_all(), which is deprecated, and
related code.
Bug: webrtc:9352
Change-Id: Ib7de5931c900eaf1023ecf3046f560feaaeec8ef
Reviewed-on: https://webrtc-review.googlesource.com/85347
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23743}
When GetSvcConfig returned fewer spatial layers than the number
statically configured from the test, we would crash on a SIGFPE.
This is not a problem in the production code, since there we
reset the encoder with the correct number of spatial layers
whenever the resolution changes.
Bug: None
Change-Id: I339e4a3c0fa993c7c649533c0eae71e1314194e7
Reviewed-on: https://webrtc-review.googlesource.com/85374
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23741}
Added distinction between number of configured and number of actively
encoded spatial layers and include number of actively encoded spatial
layers in ssData. Modified layer_filtering_transport.cc test to
parse from the RTP header and use the number of actively encoded
spatial layers for filtering spatial video layers.
Bug: webrtc:9425
Change-Id: Ic9f8895ab08b0626f9bb53a75ec33d8e7eb8706e
Reviewed-on: https://webrtc-review.googlesource.com/84243
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23716}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Bug: webrtc:5840
Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8
Reviewed-on: https://webrtc-review.googlesource.com/84743
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23715}
This reverts commit 07efe436c9.
Reason for revert: Breaks downstream project.
cricket::GetSimulcastConfig method signature has been updated.
I think you can get away with a default value for temporal_layers_supported (and then you can remove it after a few days when projects will be updated).
Original change's description:
> Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
>
> * Move SimulcastEncoderAdapter out under modules/video_coding
> * Move SimulcastRateAllocator back out to modules/video_coding/utility
> * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
> * Move any VP8 specific code - such as temporal layer bitrate budgeting -
> under codec type dependent conditionals.
> * Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
>
> Bug: webrtc:5840
> Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
> Reviewed-on: https://webrtc-review.googlesource.com/64100
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23705}
TBR=sprang@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org,hta@webrtc.org,sergio.garcia.murillo@gmail.com,titovartem@webrtc.org,agouaillard@gmail.com
Change-Id: Ic9d3b1eeaf195bb5ec2063954421f5e77866d663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5840
Reviewed-on: https://webrtc-review.googlesource.com/84760
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23710}
* Move SimulcastEncoderAdapter out under modules/video_coding
* Move SimulcastRateAllocator back out to modules/video_coding/utility
* Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility
* Move any VP8 specific code - such as temporal layer bitrate budgeting -
under codec type dependent conditionals.
* Plumb the simulcast index for H264 in the codec specific and RTP format data structures.
Bug: webrtc:5840
Change-Id: Ieced8a00e38f273c1a6cfd0f5431a87d07b8f44e
Reviewed-on: https://webrtc-review.googlesource.com/64100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23705}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
This is a reland of efc71e565e
Differs from the original cl by not widening the type of
VideoCodec::width and VideoCodec::height.
Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
>
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}
Bug: webrtc:7660
Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8
Reviewed-on: https://webrtc-review.googlesource.com/82140
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23569}
And update most internal calls to use it.
Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
This reverts commit 16e28d143a.
Reason for revert: Fix has supposedly landed upstream.
Original change's description:
> Disabling VeryLowBitrateVP9 to unblock roll.
>
> This should be re-enabled very soon since the libvpx thinks this
> is fixed upstream and is only waiting for merge.
>
> TBR=marpan@google.com
>
> Bug: webrtc:9292
> Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
> Reviewed-on: https://webrtc-review.googlesource.com/81660
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23525}
TBR=phoglund@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9292
Change-Id: I995953070536e8ee3540e7c30bc11dc1200e0463
Reviewed-on: https://webrtc-review.googlesource.com/82200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23552}
This should be re-enabled very soon since the libvpx thinks this
is fixed upstream and is only waiting for merge.
TBR=marpan@google.com
Bug: webrtc:9292
Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
Reviewed-on: https://webrtc-review.googlesource.com/81660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23525}
This is a reland of 3409cfa378
Needed to change RtpVideoStreamReceiver to stop deregistering a payload
type if two payload types refer to the same codec (which now happens,
with the packetization mode 0/1 payload types). It's not clear why this
was being done in the first place.
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
Bug: chromium:600254
Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259
Reviewed-on: https://webrtc-review.googlesource.com/78399
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23494}
This will allow exposing the interface to downstream users that
want to test VP8 simulcast. No functional changes to the tests
themselves are expected.
Bug: webrtc:9281
Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da
Reviewed-on: https://webrtc-review.googlesource.com/77361
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23469}
Use new frame dropping mode - FULL_SUPERFRAME_DROP - in VP9 encoder and
configure it to drop entire superframe if any layer is overshooting.
Bug: none
Change-Id: Ie22ed5c175e530bcce365d40cba0d10cb608ad4f
Reviewed-on: https://webrtc-review.googlesource.com/79622
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23447}
It should be the responsibility of the fixture user to provide the exact
codecs that should be tested instead. This reduces the coupling between
the test fixture and the codec instantiation.
Bug: webrtc:9317
Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b
Reviewed-on: https://webrtc-review.googlesource.com/79147
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23425}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
In flexible mode, use VP9E_GET_SVC_REF_FRAME_CONFIG to get indices of
reference frame buffers and buffers update by encoded frame.
Set inter_pic_predicted to true only if encoder actually used temporal
prediction.
Bug: webrtc:9244, webrtc:9270
Change-Id: I4e439abeab9e063d50abdcefc59bf58d6596ea6c
Reviewed-on: https://webrtc-review.googlesource.com/74780
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Cr-Commit-Position: refs/heads/master@{#23410}
Thus we don't need to initialize new members added to the structure
in the future.
Bug: None
Change-Id: Id9f5b127c224660f3016973261045b4231a617c1
Reviewed-on: https://webrtc-review.googlesource.com/79080
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23404}
This is needed for downstream users of the impl, as we currently pull
in Chromium specifics in the android_codec_factory_helper. Further,
the downstream users should explicitly supply their own factories
if they do not want to use the internal ones.
Bug: None
Change-Id: Ia7b01a66aadaba3d5accf44e5ca38e1a319e4e34
Reviewed-on: https://webrtc-review.googlesource.com/78420
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23390}
This reverts commit 3409cfa378.
Reason for revert: Broke WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsH264 on Windows 7/10 bots
Original change's description:
> Start supporting H264 packetization mode 0.
>
> The work was already done to support it, but it wasn't being negotiated
> in SDP.
>
> This means we'll now see 8 H264 payload types instead of 4; one for each
> combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
> This could be problematic in the future, since we're starting to run
> out of dynamic payload types (using 25 of 32).
>
> Bug: chromium:600254
> Change-Id: Ief2340db77c796f12980445b547b87e939170fae
> Reviewed-on: https://webrtc-review.googlesource.com/77264
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23372}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,sprang@webrtc.org
Change-Id: I2f2a2b4ca20ba883764cd5265911e1453d3df66e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:600254
Reviewed-on: https://webrtc-review.googlesource.com/78398
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23374}
The work was already done to support it, but it wasn't being negotiated
in SDP.
This means we'll now see 8 H264 payload types instead of 4; one for each
combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX.
This could be problematic in the future, since we're starting to run
out of dynamic payload types (using 25 of 32).
Bug: chromium:600254
Change-Id: Ief2340db77c796f12980445b547b87e939170fae
Reviewed-on: https://webrtc-review.googlesource.com/77264
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23372}
These files are required when implementing tests based on the test fixture,
and should be exposed as part of the test api.
This CL also removes a usage of stringstream and fixes some chromium-style
lint issues.
Bug: webrtc:8982, webrtc:163
Change-Id: I132aea0da79a79587887f21897236fc9802b7574
Reviewed-on: https://webrtc-review.googlesource.com/74586
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23346}
Follow up to https://webrtc-review.googlesource.com/c/src/+/39511,
which introduced a new Decode method, without the
RTPFragmentationHeader argument, and deprecated the old method.
Bug: webrtc:6471
Change-Id: Icd3c536ebedd4e3c2d57fdb4d6e078d6ff1de5b6
Reviewed-on: https://webrtc-review.googlesource.com/75180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23339}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
- Limit framerate by dropping frames before encoding.
- The max framerate at screen sharing is set to 5fps.
Bug: webrtc:9261
Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760
Reviewed-on: https://webrtc-review.googlesource.com/76921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23316}
- Remove referencing control from encoder wrapper. Use fixed temporal
prediction structure.
- Remove flexible mode from encoder wrapper. It only worked with
referencing control which this CL removes.
- Remove external framerate/bitrate controller. Keep codec's internal
frame dropping enabled at screen sharing.
- Use GetSvcConfig() to configure layering.
Bug: webrtc:9261
Change-Id: I355baa6aab7b98ac5028b3851d1f8ccc82a308e0
Reviewed-on: https://webrtc-review.googlesource.com/76801
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23311}
After https://webrtc-review.googlesource.com/c/src/+/70740, we are
creating/destroying the codecs on a task queue in the VideoStreamEncoder. This
CL updates the VideoCodecTest to do the same.
Also, this CL switches from manually Wait()'ing on the task queue to using
TaskQueueForTest::SendTask.
Bug: None
Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e
Reviewed-on: https://webrtc-review.googlesource.com/76800
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23257}
This CL simply renames the test cases that were not renamed in
prior CLs.
Bug: None
Change-Id: If616eb823e1453bc92ba037722b77a219d54409c
Reviewed-on: https://webrtc-review.googlesource.com/76780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23240}
- Two quality layers (same resolution, different bitrate).
- Max bitrate of low layer is limited to 200kbps. The choice of the
limit is driven by VP8 screen sharing which limits max bitrate of low
temporal layer to 200kbps. Using the same value for VP9 guarantees
that there will be no regressions for participants with limited
bandwidth.
- Max bitrate of high layer is limited to 500kbps. According to test
results this value is enough to get up to +5dB higher PSNR than VP8
SS provides on 1.2Mbps (max total bitrate for VP8 SS) link.
- Max total sent bitrate is limited to 700kbps. It is 500kbps lower
than that in VP8 SS (1200kbps).
Bug: webrtc:9261
Change-Id: I7919cc3933064664567c39e380a44cad0c65f1e8
Reviewed-on: https://webrtc-review.googlesource.com/76380
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23226}
Changed minimum bit rate threshold formula to raise the minimum bit rate
at which 720p video is presented in VP9 SVC to ensure that the video
quality for VP9 SVC is the same or better than VP8 SIM. The minimum bit
rate threshold values for lower resolutions remain largely unchanged.
Also changed maximum bit rate threshold formula to lower the maximum
bit rate for low resolutions (e.g., 180p) in order to ensure higher
frame rates when downlink bit rates are very low (e.g., < 100 kbps).
Bug: webrtc:9242
Change-Id: I8f9c76c9188b98f3fd40a608551b576b0c3b8f34
Reviewed-on: https://webrtc-review.googlesource.com/75244
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23218}
This removes second reference for frame 3 in GOF predefined for 3
temporal layers since encoder never use that reference.
Bug: webrtc:9245
Change-Id: I6fbdbe7d3c753dda7fbcfcbd05f3530f70f80728
Reviewed-on: https://webrtc-review.googlesource.com/74705
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23193}
Before this CL, there would be an out-of-bounds write in the ByteBuffer
copying when a decoded frame had height != sliceHeight.
Bug: webrtc:9194
Change-Id: Ibb80e5555e8f00d9e1fd4cb8a73f5e4ccd5a0b81
Tested: 640x360 loopback with eglContext == null in AppRTCMobile on Pixel.
Reviewed-on: https://webrtc-review.googlesource.com/74120
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23184}
Intend to delete in a later cl.
Bug: webrtc:6471
Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a
Reviewed-on: https://webrtc-review.googlesource.com/39511
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23162}
The code changes in this CL configure VP9 SVC to drop a superframe when
the spatial base layer is dropped and to not drop upper spatial layers
when the spatial base layer is not dropped. The changes are effective in
non-flexible mode when codec_.mode == kRealtimeVideo and
number of spatial layers > 1.
Bug: none
Change-Id: I27481b78f733cfc6c007d1ad9f45d69263853149
Reviewed-on: https://webrtc-review.googlesource.com/74261
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23127}
This CL creates a test fixture for the videoprocessor integration tests
and exposes it as part of the public API. It also rewrites the current
versions of the tests to build on this new paradigm. The motivation for
this is to easily allow projects that build on top of webrtc to add
integration-level tests for their own custom codec implementations in a
way that does not link them too tightly to the internal implementations
of said tests.
Bug: None
Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2
Reviewed-on: https://webrtc-review.googlesource.com/72481
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23118}
Request key frame when upper spatial layer is enabled dynamically
and inter-layer prediction is disabled or limited to key pictures.
This is needed to force encoder to produce RTP compatible bitstream
where temporal prediction is limited to the same spatial layer.
Bug: webrtc:9217
Change-Id: I4fc1e3f067689ba7b5c6bd1f5af922a0637f03d7
Reviewed-on: https://webrtc-review.googlesource.com/73580
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23102}
Fill drops with last decoded frame to make them look like freeze at
playback and to keep decoded spatial layers in sync.
Bug: none
Change-Id: I65f7c21100985c22932a1edd441b6c724833c11e
Reviewed-on: https://webrtc-review.googlesource.com/73685
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23076}
If frame of current layer was dropped, pass base frame to decoder if
non_ref_for_inter_layer_pred is set to true.
Bug: none
Change-Id: If7bf5220b74f424106edf74867c9afa8cc2b1ec5
Reviewed-on: https://webrtc-review.googlesource.com/73440
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23074}
This allows to control inter-layer prediction at encoding VP9 SVC.
There are three options:
1. Disabled.
2. Enabled for all pictures.
3. Enabled for key pictures, disabled for others.
Inter-layer prediction is enabled for all pictures by default.
Bug: none
Change-Id: I49fe43d8744c92bec349d815100ba158519f0664
Reviewed-on: https://webrtc-review.googlesource.com/71500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23049}
For consistency with the VP9 RTP spec which uses term "picture" for set
of frames which belong to the same time instance.
Bug: none
Change-Id: I30e92d5debb008feb58f770b63fe10c2e0029267
Reviewed-on: https://webrtc-review.googlesource.com/72180
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23040}
This deletes the resilienceOn flag in VideoCodecVP8 and VideoCodecVP9.
Instead, the implementations of VP8 and VP9 set resilience mode
internally, based on the configuration of temporal and spatial layers.
The nack_enabled argument to VideoCodecInitializer::SetupCodec becomes
unused with this cl. In a followup, it will be deleted, together with
the corresponding argument to VideoStreamEncoder methods.
An applications which really wants to configure resilience differently
can do that by injecting an EncoderFactory with encoders behaving
as desired.
Bug: webrtc:8830
Change-Id: I9990faf07d3e95c0fb4a56fcc9a56c2005b4a6fa
Reviewed-on: https://webrtc-review.googlesource.com/71380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23025}
This converts the reserved bit in VP9 RTP payload descriptor into the
flag which indicates whether current frame can be used for prediction
of next spatial layer or not.
VP9 encoder wrapper sets non_ref_for_inter_layer_pred=false for all
frames for now.
Bug: none
Change-Id: I32f68868686475905fb09173cffd2b6e1bedcb7c
Reviewed-on: https://webrtc-review.googlesource.com/71080
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23010}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
We only support on (formely kResilientStream) and off (formely
kResilienceOff). The third mode, kResilientFrames, was not
implemented.
Bug: None
Change-Id: Ida82f6a33eda9d943ea70bc8ae4e6bddb720b0e8
Reviewed-on: https://webrtc-review.googlesource.com/71481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22984}
This fixes misprint in the code which calculates target bitrate of a
VP9 spatial layer where "-" was used instead of "+".
Bug: none
Change-Id: I17d76a84d00e453c055c068968d7b276e9c23f51
Reviewed-on: https://webrtc-review.googlesource.com/71663
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22974}
Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
Bug: webrtc:8982
Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
Reviewed-on: https://webrtc-review.googlesource.com/64143
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22916}
With rounding to the nearest the result can exceed the allocated
bitrate.
Bug: none
Change-Id: I0260a1640a1454951ca8e48fd447e047ef0271ee
Reviewed-on: https://webrtc-review.googlesource.com/69982
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22879}
This fixes inconsistency in names of variables and fields which
represent spatial/temporal index of layer:
simulcast_svc_idx -> spatial_idx
spatial_layer_idx -> spatial_idx
temporal_layer_idx -> temporal_idx
Also, this adds printing of spatial/temporal index and target bitrate
to RD report.
Bug: none
Change-Id: Ic4dfdadc57a1577bb3d35d1782a152a9dbef0280
Reviewed-on: https://webrtc-review.googlesource.com/69981
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22869}
Adding SVC rate allocator and layering configurator caused regression
for VP9 non-SVC senders. SVC bitrate limits, which were supposed to
be used only when spatial layering is enabled, are applied when
encoding single spatial layer. E.g. for VP9 360p sender maximum bitrate
is limited to 500kbps.
This fixes the regression. If sender is configured to send VP9 single
layer then codec's bitrate limits are applied to this layer.
Bug: webrtc:9151, chromium:831093
Change-Id: Ia1ae4087155ad7917a3443304a21532f1e68ea65
Reviewed-on: https://webrtc-review.googlesource.com/69813
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22862}
This makes it easier to add new test cases without modifying the actual test class.
Bug: None
Change-Id: I48e4f14e26cd6610678ffb07ce9fd56e6bc1ac4e
Reviewed-on: https://webrtc-review.googlesource.com/69600
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22840}
OnRatesUpdated() is called every time the bitrate estimate, or once per
second. However, since we don't want to reconfigure libvpx too often,
just in case it interferes with the rate controller, so
ScreenshareLayers contains a boolean |bitrate_update_| which indicate
if the configuration should be updated on a call to
UpdateConfiguration().
However, it two rate updates happened between two frames, the first of
which changes the rates and second one does not, |bitrate_update_| will
be reset to false and the encoder won't get the desired config.
This CL makes sure we update the configuration iff the rate has changed
at any time since the last call to UpdateConfiguration().
Bug: webrtc:9012
Change-Id: I62af36cffe20ecb7d3f403b3eb11f23a9692d719
Reviewed-on: https://webrtc-review.googlesource.com/69040
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22826}
Set the flag based on coded length of buffered frame which is reset
after picture is encoded and, thus, is equal to zero when encoder
delivers first frame of next picture.
Before this change first_frame_in_picture was set based on index of
spatial layer of encoded frame. This is not right anymore since encoder
can drop base layer but deliver upper layers.
Bug: chromium:828350
Change-Id: I12c7534240de8bc4905f04ff368cc3704720a70b
Reviewed-on: https://webrtc-review.googlesource.com/68561
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22805}
It takes some time for rate controller to adapt to content. Quality of first
frames is usually worse than quality of following frames. It makes sense to
exclude first frames from analysis and, thus, avoid negative affect of this
interval on overall results.
Bug: none
Change-Id: Ib0a258889750cf794c7d6fdff26af958f7bbe48a
Reviewed-on: https://webrtc-review.googlesource.com/66100
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22782}
In order to handle per-layer frame dropping both VP9 encoder wrapper
and RTP packetizer were modified.
- Encoder wrapper buffers last encoded frame and passes it to
packetizer after frame of next layer is encoded or encoding of
superframe is finished.
- Encoder wrapper sets end_of_superframe flag on last encoded frame of
superframe before passing it to packetizer.
- If end_of_superframe is True then packetizer sets marker bit on last
packet of frame.
Bug: webrtc:9066
Change-Id: I1d45319fbe6bc63d01721ea67bfb7440d4c29275
Reviewed-on: https://webrtc-review.googlesource.com/65540
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22722}
Measure time spent in frame encode callback, accumulate it for layers
and subtract it from measured encode time of next layer frame.
Bug: none
Change-Id: Ifc3baae2f9a49913a55a7de2de9507102edd0295
Reviewed-on: https://webrtc-review.googlesource.com/65981
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22720}
The configurator decides number of spatial layers, their resolution
and bitrate thresholds based on given input resolution and maximum
number of spatial layers.
The allocator distributes available bitrate across spatial and
temporal layers. If there is not enough bitrate to provide acceptable
quality for all spatial layers allocator disables enhancement layers
one by one until the condition is met or number of layers is reduced
to one.
VP9 SVC related unit tests have been updated. Input resolution and
bitrate in these tests have been increased to the level enough to
provide desirable number of spatial layers.
Bug: webrtc:8518
Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b
Reviewed-on: https://webrtc-review.googlesource.com/60340
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22672}
There is no need to use real video as input for encoder in unit tests.
Using generator simplifies testing on mobile devices (no need to upload
files to device).
Bug: none
Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398
Reviewed-on: https://webrtc-review.googlesource.com/64526
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22648}
Build superframe out of the nearest non-dropped base layer and current layer.
Bug: none
Change-Id: I26720ea6de44f27046208b220d03942cd2a3d6c7
Reviewed-on: https://webrtc-review.googlesource.com/64921
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22631}
It would be nice to also delete the fields from CodecSpecificInfo,
but those fields are used on the receive side.
Bug: webrtc:8830
Change-Id: I1a3f13ea2c024cbd73b33fd9dd58e531d3576a55
Reviewed-on: https://webrtc-review.googlesource.com/64780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22625}
Have not figured out why this metric regressed, but submitting
this CL now to unblock Chromium roll into WebRTC.
Bug: webrtc:9057
Change-Id: I808ad194e1c9107d644a25502a55a7c6fddca7aa
Reviewed-on: https://webrtc-review.googlesource.com/64527
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22600}
As the rate allocation has been moved into entirely into
SimulcastRateAllocator, and the listeners are thus no longer needed,
this class doesn't fill any other purpose than to determine if
ScreenshareLayers or TemporalLayers should be created for a given
simulcast stream. This can however be done just from looking at the
VideoCodec instance, so changing this into a static factory method.
Due to dependencies from upstream projects, keep the class name and
field in VideoCodec around for now.
Bug: webrtc:9012
Change-Id: I028fe6b2a19e0d16b35956cc2df01dcf5bfa7979
Reviewed-on: https://webrtc-review.googlesource.com/63264
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22529}
This CL moves all temporal layer rate allocation from
DefaultTemporalLayers and ScreenshareLayers into SimulcastRateAllocator.
This means we don't need an extra call-out to the TemporalLayers
interface to get the last allocation, which simplifies the code path a
lot.
It also paves the wave for removing the TemporalLayersFactory interface
(in a separate cl), which will further simplify the ownership model.
Bug: webrtc:9012
Change-Id: I6540b1848efa1a136dce449f13902ad479d5ee37
Reviewed-on: https://webrtc-review.googlesource.com/62420
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22502}
It previously owned only the picture id and only in the
WebRTC-VP8-Forced-Fallback-Encoder-v2 experiment.
Moving responsibility to PayloadRouter ensures that both
picture id and tl0 idx are continuous over codec changes,
as required by the specs for VP8 and VP9 over RTP.
Bug: webrtc:8830
Change-Id: Ie77356dfec6d1e372b6970189e4c3888451920e6
Reviewed-on: https://webrtc-review.googlesource.com/61640
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22448}
This work is in preparation for refactoring the TemporalLayers api.
Bug: webrtc:9012
Change-Id: I01908ee034fb79996e687ff72d10178acf102321
Reviewed-on: https://webrtc-review.googlesource.com/61781
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22445}
For the buffering of |input_frames_|, we assume that frames
are ordered per simulcast layer but we make no assumptions
between layers.
For SVC, we still assume ordering of encode callbacks for
the spatial layers. If we ever add async codecs that support SVC,
they should still obey this assumption.
Bug: webrtc:8448
Change-Id: I4ebb0c1e1d0eef41d850ed5b92aacc79d0a11137
Reviewed-on: https://webrtc-review.googlesource.com/60801
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22380}
* Add support for SimulcastEncoderAdapter wrapping of encoder.
* Store input frame timestamps out-of-band, so we don't need to keep
a raw VideoFrame around just for it's timestamp.
* Store current frame rate in |framerate_fps_|, instead of in
codec settings struct.
* Add some comments and reorder some data members.
* Explicitly include VideoBitrateAllocator.
* Change type of |input_frames_|, to avoid one layer of indirection.
* Move VideoProcessor::CalculateFrameQuality to anonymous namespace.
This change should have no functional implications.
Bug: webrtc:8448
Change-Id: I10c140eeda750d9bd37bfb6cb1e8acb401fb91d3
Reviewed-on: https://webrtc-review.googlesource.com/60520
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22346}
Previously, the only user of this code was the
VideoProcessorIntegrationTest. We have now changed that
test to directly calculate image quality metrics using libyuv,
similar to how the full stack tests and browser tests work.
Bug: webrtc:8448
Change-Id: Ia7a607d7ddc37741fba76d56aa7297851ffa1c6b
Reviewed-on: https://webrtc-review.googlesource.com/43760
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22341}
It holds the same information as codecType, but in different format.
Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
These tests cannot run on simulators but should be enabled on real device
bots in order to catch regressions or crashes in the iOS codecs.
Bug: webrtc:8950
Change-Id: I8e877aa4368683073fdb4586cd6f4add4a1284ad
Reviewed-on: https://webrtc-review.googlesource.com/59040
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22283}
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.
Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
Both simulcast stream and spatial layer can be described with the same
set of parameters. There is no need in two separate definitions.
1. Original definition of SpatialLayer is removed.
2. SimulcastStream is renamed to SpatialLayer.
3. SimulcastStream is equated to SpatialLayer using typedef.
Bug: webrtc:8518
Change-Id: I90761952b032a1b71fc4bba11f74a6daaf58880a
Reviewed-on: https://webrtc-review.googlesource.com/57102
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22272}
This fixes the issue when Init() with correct codec settings fails
because preceding Init() was called with wrong settings.
Bug: webrtc:8969
Change-Id: I50e618af6266ef593942fda27839c7c01e8717ae
Reviewed-on: https://webrtc-review.googlesource.com/59382
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22271}
This CL adds a lock to stashed_images_ in MultiplexEncoderAdapter.
Without lock, it is possible that different threads acts on
stashed_images_ simultaneously and leads to crash.
Bug: webrtc:8965
Change-Id: I887861092d185c3bd6047eb529d8c1cf57fa4648
Reviewed-on: https://webrtc-review.googlesource.com/59260
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22261}
This reverts commit e27e0aca94.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."
>
> This reverts commit d2ed0a4c9e.
>
> Reason for revert: Breaks downstream projects.
>
> Original change's description:
> > Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers.
> >
> > temporal_layer_thresholds_bps served only one purpose: its size was used
> > to infer number of temporal layers. I replaced it with num_temporal_layers,
> > which does what is says.
> >
> > The practical reason for this change is the need to have possibility to
> > distinguish between cases when VP9 SVC temporal layering was/not set
> > through field trial. That was not possible with
> > temporal_layer_thresholds_bps[] because empty vector means 1 temporal
> > layer.
> >
> > Bug: webrtc:8518
> > Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
> > Reviewed-on: https://webrtc-review.googlesource.com/58084
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22230}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
>
> Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8518
> Reviewed-on: https://webrtc-review.googlesource.com/58902
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22234}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,ssilkin@webrtc.org
Change-Id: I1900c6b845b9baa9430fb72c3f4e7f2a44b3a8b1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8518
Reviewed-on: https://webrtc-review.googlesource.com/59160
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22256}
* Do not simulate freeze in decoded output file when frames have been dropped.
* Add more DCHECKs and consts.
* Remove unused members |num_encoded_frames_| and |num_decoded_frames_|.
* Move SdpVideoFormat conversion to TestConfig.
Bug: webrtc:8448
Change-Id: Ia879141f36dc23427cd1abcaa66716656fbaac2a
Reviewed-on: https://webrtc-review.googlesource.com/56802
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22239}
This reverts commit d2ed0a4c9e.
Reason for revert: Breaks downstream projects.
Original change's description:
> Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers.
>
> temporal_layer_thresholds_bps served only one purpose: its size was used
> to infer number of temporal layers. I replaced it with num_temporal_layers,
> which does what is says.
>
> The practical reason for this change is the need to have possibility to
> distinguish between cases when VP9 SVC temporal layering was/not set
> through field trial. That was not possible with
> temporal_layer_thresholds_bps[] because empty vector means 1 temporal
> layer.
>
> Bug: webrtc:8518
> Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
> Reviewed-on: https://webrtc-review.googlesource.com/58084
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22230}
TBR=sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Ic2940f7f78a74312170940d51ad8967cde8ad42f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8518
Reviewed-on: https://webrtc-review.googlesource.com/58902
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22234}
temporal_layer_thresholds_bps served only one purpose: its size was used
to infer number of temporal layers. I replaced it with num_temporal_layers,
which does what is says.
The practical reason for this change is the need to have possibility to
distinguish between cases when VP9 SVC temporal layering was/not set
through field trial. That was not possible with
temporal_layer_thresholds_bps[] because empty vector means 1 temporal
layer.
Bug: webrtc:8518
Change-Id: I275ec3a8c74e8ba409eb049878199f132a20ec51
Reviewed-on: https://webrtc-review.googlesource.com/58084
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22230}
Uploading of the file to device causes timeouts. I removed it from
resources for iOS and Android builds and disabled the test that used
the file since we don't really need to run it on test bots.
Bug: webrtc:8936
Change-Id: Ia5e04c4630544eca8e56826c9e89c9c9f4dcb600
Reviewed-on: https://webrtc-review.googlesource.com/58090
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22222}
This CL fixes the bug that we must have some padding for H264 codec
when used in Multiplex encoder/decoder, otherwise H264 decoder will
crash.
And this CL fixes a bug that potential infinite loop exists in
MultiplexEncoderAdapter
Bug: webrtc:8921
Change-Id: I4124579c31ee69f72e29d118378aa1f8b3f05eb4
Reviewed-on: https://webrtc-review.googlesource.com/56960
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22175}
Overriding implementations of VideoEncoder::GetScalingSettings that
want to enable quality scaling must now provide the thresholds.
Bug: webrtc:8830
Change-Id: I75c47cb56ac1b9cf77401684980b3167e485f51c
Reviewed-on: https://webrtc-review.googlesource.com/46622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22172}
This script is for running on device tests in parallel.
BUG=webrtc:8448
NOTRY=TRUE
Change-Id: I6b13f76223653ddb6ec999613ef66ac4f82d8567
Reviewed-on: https://webrtc-review.googlesource.com/55561
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22117}
Slicing, aggregation and analysis has been moved to Stats class.
Data of all spatial layers is stored in single Stats object.
Bug: webrtc:8524
Change-Id: Ic9a64859a36a1ccda661942a201cdeeed470686a
Reviewed-on: https://webrtc-review.googlesource.com/50301
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22094}
Apparently left over since 2015 refactorings, see cl
https://codereview.webrtc.org/1417283007
Bug: webrtc:5095
Change-Id: I899f2c018d1906b4336b2e80b511f7398bac4947
Reviewed-on: https://webrtc-review.googlesource.com/53200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22055}
Prior to this change, the VideoProcessor was run on the main thread
in the unit tests. Using a TaskQueue there instead, we can be
stricter in the thread checks.
Bug: webrtc:8524
Change-Id: Ice7b68f7344fc52801dff7a98cbc219b7231bfbc
Reviewed-on: https://webrtc-review.googlesource.com/48921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21942}
Accessing this method from the test thread is illegal,
but doesn't always fail.
Bug: webrtc:8524
Change-Id: Ie0e84cc2fb63268fb6d7cbf0c3a58cb35312c16b
Reviewed-on: https://webrtc-review.googlesource.com/49061
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21930}
This is a reland of 8fb22e71ee.
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=sprang@webrtc.org
Bug: webrtc:8653
Change-Id: I32fa92649f3ff40b1e364f880040e52ae698f74d
Reviewed-on: https://webrtc-review.googlesource.com/46860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21918}
This helps separate concerns, so that the VideoProcessorIntegrationTest
is almost oblivious to the fact that it needs to connect to the JVM
to get the Android HW codecs.
Bug: webrtc:8448
Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8
Reviewed-on: https://webrtc-review.googlesource.com/47384
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21890}
The VideoCodecTest class is a fixture base class for the
libvpx-VP8, libvpx-VP9, and OpenH264 unit tests. It is unrelated
to the VideoProcessor tests, which we colloquially refer to as
the "codec test".
This rename is thus to reduce this confusion. It should have no
functional impact.
Bug: webrtc:8448
Change-Id: If73443bda5df0f29a71ce6ce069ac128795ff0ad
Reviewed-on: https://webrtc-review.googlesource.com/47160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21867}
This can be used to wrap Objective-C components in C++ classes, so users
can use the WebRTC C++ API directly together with the iOS specific
components provided by our SDK.
Bug: webrtc:8832
Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8
Reviewed-on: https://webrtc-review.googlesource.com/46162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21850}
Encoded frames are preserved and decoded after all layers are
encoded.
Each spatial layer is decoded with separate decoder.
For quality evaluation of lowres layers original input frame is
downscaled with bilinear interpolation.
Encoded and decoded frames are dumped into separate files.
For async codecs encoded frames are passed to decoder in encode
callback, as before.
Bug: webrtc:8524
Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1
Reviewed-on: https://webrtc-review.googlesource.com/43381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21844}
This reverts commit 8fb22e71ee.
Reason for revert: breaks downstream projects
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,shampson@webrtc.org
Change-Id: If14074a7fc56c83b75584d8e9a6a913a40514bad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/46840
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21832}
Due to libvpx we were restricted to always turning the low simulcast
stream on, or else the encoder would always label the active streams'
encoded frames as key frames. Now that libvpx has been updated and
rolled in, this change updates tests to reflect that it is working.
Bug: webrtc:8653
Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
Reviewed-on: https://webrtc-review.googlesource.com/46340
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21831}
This fix the mismatch of resolution VP9 wrapper set for coded frame with
its actual resolution.
Bug: webm:1485, webrtc:5749
Change-Id: Ie1225d8f3a3d00e66229a1a79858d0a89b3d5fae
Reviewed-on: https://webrtc-review.googlesource.com/46040
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21819}
This is a reland of bbdabe50db.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511
Reviewed-on: https://webrtc-review.googlesource.com/45820
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21794}
This reverts commit 4954a77cf8.
Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(
Original change's description:
> Reland "Rename stereo video codec to multiplex"
>
> This is a reland of bbdabe50db.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
>
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
>
> TBR=niklas.enbom@webrtc.org
>
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
This is a reland of bbdabe50db.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
This reverts commit bbdabe50db.
Reason for revert: This breaks the internal build.
Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}
TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
This reverts commit 4dc891f5e3.
Reason for revert: Reverting this CL to make it possible to revert https://webrtc-review.googlesource.com/c/src/+/43242
Original change's description:
> Add unit tests covering MultiplexImageComponent
>
> This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
> adds unit test coverage in TestMultiplexAdapter for these structs.
>
> Bug: webrtc:7671
> Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
> Reviewed-on: https://webrtc-review.googlesource.com/44081
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Qiang Chen <qiangchen@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#21770}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I9cce6ed5f2990a2f443e04a9e5913cbd296242e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44341
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21773}
This CL changes some types in MultiplexImage and MultiplexImageComponent. Also,
adds unit test coverage in TestMultiplexAdapter for these structs.
Bug: webrtc:7671
Change-Id: I832d0466dc67d3b6b7fa0d3fb76f02c0190e474f
Reviewed-on: https://webrtc-review.googlesource.com/44081
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21770}
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.
Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
This is in preparation for
https://webrtc-review.googlesource.com/c/src/+/36340
With these changes we can avoid some strange #ifdefs in the code
that uses temporal layers.
Bug: webrtc:7925
Change-Id: I472210738ccc9f73812b8863951befeabec56f15
Reviewed-on: https://webrtc-review.googlesource.com/41280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21759}
This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481.
With the new approach we can just use the generic RTP packetizer to pass frames
over the wire as the specific info is contained within the bitstream. This makes
the new codec more modular and reduces its footprint.
Bug: webrtc:7671
Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552
Reviewed-on: https://webrtc-review.googlesource.com/43220
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21753}
This reverts commit d756fd06fe.
Original change's description:
> Revert "Wrap Alpha and YUV frame into one EncodedImage for transmission"
>
> This reverts commit 5670c86aec.
>
> Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy.
>
> Original change's description:
> > Wrap Alpha and YUV frame into one EncodedImage for transmission
> >
> > With alpha channel, we observe the artifacts on the receiver side, and
> > the reason is that when YUV channel has a key frame, it gives frame_buffer2
> > a chance to drop some previous frames. Then it is possible that some alpha
> > frames got dropped, which break the alpha frame dependence chain.
> >
> > In this CL, we pack the YUV frame and alpha encoded frame together as one
> > entity to solve the issue.
> >
> > Bug: webrtc:8773
> > Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
> > Reviewed-on: https://webrtc-review.googlesource.com/38481
> > Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21737}
>
> TBR=qiangchen@chromium.org,emircan@webrtc.org
>
> Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8773
> Reviewed-on: https://webrtc-review.googlesource.com/43260
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21739}
TBR=deadbeef@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I0d64b7e7a62e4f35aa012270d3826a23b3fb2337
Bug: webrtc:8773
Reviewed-on: https://webrtc-review.googlesource.com/43440
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21749}
Each simulcast stream requires dedicated decoder for decoding. SVC
can be decoded by single decoder. But in prod each receiver has its
decoder. We want to replicate this and also use one decoder per
spatial layer.
Also we create one frame writer per simulcast/spatial layer to dump
encoded/decoded frames of different layers to separate files.
Note that videoprocessor is still initialized with single
decoder/writer. It will be updated in next CL and start using
separate decoder/writer per layer.
Bug: webrtc:8524
Change-Id: I3bb3de77f97d51138b8b7675dd01bc281a078b2f
Reviewed-on: https://webrtc-review.googlesource.com/43280
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21744}
These parameters allow to configure number of simulcast/spatial layers
in video codec tests.
Bug: webrtc:8524
Change-Id: Iad1332732758a8297abcf740c24c483e5fccec9a
Reviewed-on: https://webrtc-review.googlesource.com/43020
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21741}
This reverts commit 5670c86aec.
Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy.
Original change's description:
> Wrap Alpha and YUV frame into one EncodedImage for transmission
>
> With alpha channel, we observe the artifacts on the receiver side, and
> the reason is that when YUV channel has a key frame, it gives frame_buffer2
> a chance to drop some previous frames. Then it is possible that some alpha
> frames got dropped, which break the alpha frame dependence chain.
>
> In this CL, we pack the YUV frame and alpha encoded frame together as one
> entity to solve the issue.
>
> Bug: webrtc:8773
> Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
> Reviewed-on: https://webrtc-review.googlesource.com/38481
> Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21737}
TBR=qiangchen@chromium.org,emircan@webrtc.org
Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8773
Reviewed-on: https://webrtc-review.googlesource.com/43260
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21739}
With alpha channel, we observe the artifacts on the receiver side, and
the reason is that when YUV channel has a key frame, it gives frame_buffer2
a chance to drop some previous frames. Then it is possible that some alpha
frames got dropped, which break the alpha frame dependence chain.
In this CL, we pack the YUV frame and alpha encoded frame together as one
entity to solve the issue.
Bug: webrtc:8773
Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7
Reviewed-on: https://webrtc-review.googlesource.com/38481
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21737}
Try to use frame timestamps first if they look reasonable, otherwise
use realtime clock.
Also, lower limit from 90% of target to 85%.
Bug: webrtc:4172, chromium:802290
Change-Id: Iad489be7c7cf637345be4795e5089936ab9fab07
Reviewed-on: https://webrtc-review.googlesource.com/41041
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21729}
This feature is not needed in video codec testing framework. In WebRTC
video codecs never deal with packet loss. Packet loss is handled by
jitter buffer which prevents passing of incomplete frames to decoder.
Bug: webrtc:8768
Change-Id: I211cf51d913bec6a1f935e30691661d428ebd3b6
Reviewed-on: https://webrtc-review.googlesource.com/40740
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21722}
No stats is logged in this format any longer.
Bug: none
Change-Id: I5f91e93636b6d03ebd91c3b2518857275fb94de7
Reviewed-on: https://webrtc-review.googlesource.com/40700
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21690}
This is a reland of 18c4261339
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=sprang@webrtc.org,stefan@webrtc.org,deadbeef@webrtc.org
Bug: webrtc:8630
Change-Id: Ib3df6f9b7158bff362a7ec66fc57e368682c5846
Reviewed-on: https://webrtc-review.googlesource.com/40980
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21688}
This is a reland of 1880c7162b
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org, stefan@webrtc.org
Bug: webrtc:8524
Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae
Reviewed-on: https://webrtc-review.googlesource.com/40202
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21668}
This reverts commit 1880c7162b.
Reason for revert: breaks internal tests
Original change's description:
> Updated analysis in videoprocessor.
>
> - Run analysis after all frames are processed. Before part of it was
> done at bitrate change points;
> - Analysis is done for whole stream as well as for each rate update
> interval;
> - Changed units from number of frames to time units for some metrics
> and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
> 'time to reach target bitrate, sec';
> - Changed data type of FrameStatistic::max_nalu_length (renamed to
> max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
> use such advanced data type in such low level data structure.
>
> Bug: webrtc:8524
> Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
> Reviewed-on: https://webrtc-review.googlesource.com/31901
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21653}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,ssilkin@webrtc.org
Change-Id: Id0b7d387bbba02e71637b229aeed6f6cf012af46
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8524
Reviewed-on: https://webrtc-review.googlesource.com/40220
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21656}
This reverts commit 16beea7c90.
Reason for revert: breaks internal tests.
Original change's description:
> Set encodedWidth/encodedHeight to actual coded resolution.
>
> This fixes mismatch between resolution of coded frame indicated by VP9
> encoder wrapper and actual coded resolution. If internal resize is
> enabled VP9 encoder might downscale input frame when bitrate is too low
> to keep good spatial quality. Before this fix VP9 wrapper always set
> coded resolution equal to input resolution. Now it sets it to actual
> coded resolution which it reads from frame pkt.
>
> Bug: webrtc:5749
> Change-Id: I7dc8ba89947e99213a3b4c3cd4d974b662f090c4
> Reviewed-on: https://webrtc-review.googlesource.com/39661
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21651}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,ssilkin@webrtc.org
Change-Id: I32cb1271c863cf435f3fc3b465e11232cbd6a888
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:5749
Reviewed-on: https://webrtc-review.googlesource.com/40161
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21655}
- Run analysis after all frames are processed. Before part of it was
done at bitrate change points;
- Analysis is done for whole stream as well as for each rate update
interval;
- Changed units from number of frames to time units for some metrics
and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to
'time to reach target bitrate, sec';
- Changed data type of FrameStatistic::max_nalu_length (renamed to
max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to
use such advanced data type in such low level data structure.
Bug: webrtc:8524
Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f
Reviewed-on: https://webrtc-review.googlesource.com/31901
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21653}
This fixes mismatch between resolution of coded frame indicated by VP9
encoder wrapper and actual coded resolution. If internal resize is
enabled VP9 encoder might downscale input frame when bitrate is too low
to keep good spatial quality. Before this fix VP9 wrapper always set
coded resolution equal to input resolution. Now it sets it to actual
coded resolution which it reads from frame pkt.
Bug: webrtc:5749
Change-Id: I7dc8ba89947e99213a3b4c3cd4d974b662f090c4
Reviewed-on: https://webrtc-review.googlesource.com/39661
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21651}
This reverts commit 18c4261339.
Reason for revert: Broke internal tests
Original change's description:
> Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
>
> Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
>
> Bug: webrtc:8653
> Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
> Reviewed-on: https://webrtc-review.googlesource.com/37740
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21646}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,shampson@webrtc.org
Change-Id: I0aeb743cbd2e8d564aa732c937587c25a4c49b09
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/39883
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21647}
Creates VideoStreams & VideoCodec.simulcastStreams with an active field, and then allocates 0 bitrate to simulcast streams that are inactive. This turns off the encoder for specific simulcast streams.
Bug: webrtc:8653
Change-Id: Id93b03dcd8d1191a7d3300bd77882c8af96ee469
Reviewed-on: https://webrtc-review.googlesource.com/37740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21646}
This is a reland of https://webrtc-review.googlesource.com/34380
The main problem with that CL was that we used frame timestamps as basis
for frame dropping, but those might not be continuous or even populated
in some circumstances.
Additionally, I found that the bitrate was off since the encoder does
not not take the dropped frames into account, so if we drop every other
frame continiusoly, the bitrate sent will be around half of the target.
Patch set 1 is the original CL, subsequent patch sets cotains fixes.
Bug: webrtc:4172
Change-Id: I8ec8dddcebf4ce44f28dd9055cf9c46bbd68e4a6
Reviewed-on: https://webrtc-review.googlesource.com/39201
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21601}
We currently use raw jobject in our code mixed with sporadic
ScopedLocalRefFrame. This CL moves every jobject into a scoped object,
either local, global, or a parameter. Also, this CL uses the JNI
generation script to generate declaration stubs for the Java->C++
functions so that it no longer becomes possible to mistype them
without getting compilation errors.
TBR=brandt@webrtc.org
Bug: webrtc:8278,webrtc:6969
Change-Id: Ic7bac74a89c11180177d65041086d7db1cdfb516
Reviewed-on: https://webrtc-review.googlesource.com/34655
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21387}
This reverts commit 28a06b16cc.
Reason for revert: Causes some unexpected perf changes.
Original change's description:
> Smoother frame dropping when screenshare_layers limits fps
>
> Currently, when input fps is higher than the configured target fps in
> screenshare_layers, we drop frames with the help of a rate tracker using
> a one second sliding window. This is not optimal.
> For instance, given a 5fps limit and a 30fps capturer, the window will
> not be saturated until we have added the first 5 frames. Then we will
> drop all frames until the oldest one drops out, at which point we can
> immediately fill it's place. This results in quick 5 frame bursts every
> second.
>
> In addition to rate tracker, also set a limit on minimum interval
> required between input frames, based on target frame rate.
>
> Bug: webrtc:4172
> Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
> Reviewed-on: https://webrtc-review.googlesource.com/34380
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21325}
TBR=ilnik@webrtc.org,sprang@webrtc.org
Change-Id: I7ca5b0c847310dbb11dce28773082eac946c0ba4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4172
Reviewed-on: https://webrtc-review.googlesource.com/34780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21354}
Currently, when input fps is higher than the configured target fps in
screenshare_layers, we drop frames with the help of a rate tracker using
a one second sliding window. This is not optimal.
For instance, given a 5fps limit and a 30fps capturer, the window will
not be saturated until we have added the first 5 frames. Then we will
drop all frames until the oldest one drops out, at which point we can
immediately fill it's place. This results in quick 5 frame bursts every
second.
In addition to rate tracker, also set a limit on minimum interval
required between input frames, based on target frame rate.
Bug: webrtc:4172
Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
Reviewed-on: https://webrtc-review.googlesource.com/34380
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21325}
TBR=brandtr@webrtc.org,stefan@webrtc.org
Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.
Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.
Bug: webrtc:8643
Change-Id: I553cea30dcda34b753b5224f15094a1b7b70a750
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#21249}
Reviewed-on: https://webrtc-review.googlesource.com/33360
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21319}
This reverts commit 59283e4c66.
Reason for revert: This CL is preventing rolls into Chromium because it fails to compile with MSVC.
Sample error log:
[13258/43857] CXX obj/third_party/webrtc/video/video/send_statistics_proxy.obj
FAILED: obj/third_party/webrtc/video/video/send_statistics_proxy.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes @obj/third_party/webrtc/video/video/send_statistics_proxy.obj.rsp /c ../../third_party/webrtc/video/send_statistics_proxy.cc /Foobj/third_party/webrtc/video/video/send_statistics_proxy.obj /Fd"obj/third_party/webrtc/video/video_cc.pdb"
../../third_party/webrtc/video/send_statistics_proxy.cc(217): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/video/send_statistics_proxy.cc(217): warning C4267: 'initializing': conversion from 'size_t' to 'int', possible loss of data
../../third_party/webrtc/video/send_statistics_proxy.cc(632): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
Original change's description:
> googBandwidthLimitedResolution stat is not always set depending on configuration.
>
> Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
> OnEncodedImage callback.
>
> Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
> on info that is reported to SendStatisticsProxy::OnEncodedImage.
>
> Bug: webrtc:8643
> Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
> Reviewed-on: https://webrtc-review.googlesource.com/31460
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21249}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8643
Change-Id: Ib9ef55b8894ea72236a5dc1e9a839adecd401afb
Reviewed-on: https://webrtc-review.googlesource.com/33100
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21284}
