This CL fixes a few issues where the reported fraction of frames
allocated to various temporal layers could be incorrect:
* In LibvpxVp8Encoder, calling GetEncoderInfo() while not initialized,
or when first configuring with temporal layers and then without,
could trigger incorrect fps allocations.
* In VP9 when different spatial layers have different max framerates,
the layer fps should be compared to the layer with the highest
configured fps, not codec_.maxFramerate which is updated to the
current input fps on SetRates().
* In EncoderBitrateAdjuster, just warn and ignore if a layer has
non-zero bps but zero fps, rather than passing down the chain and
risk weird behavior or divide by zero.
Bug: b/152040235
Change-Id: I548fb3e099b1ec9f536a7b93313fb40c4d32e596
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171516
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30880}
this is a step towards resolving own todo: making AssembleFrame part of
the VideoRtpDepacketizer interface and replacing codec check with a
call to a virtual function.
RtpVideoStreamReceiver has access to the VideoRtpDepacketizers,
PacketBuffer - hasn't.
Bug: None
Change-Id: I83df09975c092bdb71bab270ced356d79a50683d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168056
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30833}
Also, make sure active flags are not lost in simulcast encoder adapter
which is needed in case of simulcast encoder adapter is used.
VP9 libvpx encoder currently ignores scaling setting for SVC, but libvpx
fix is incoming.
TESTED=On a manually patched chrome with singlecast-simulcast vp8 stream.
Bug: webrtc:11396
Change-Id: Ic81f014bec1bdaaf6d5d173743933e5d77d71ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169547
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30681}
The old implementation has undefined behavior in it (unaligned read of uint32_t)
Now it's closer to the reference implementation:
https://tools.ietf.org/html/rfc6386#section-20.2
Also, added some comments and named some variables to make it more clear, that the
parser actually does.
Bug: chromium:1057551
Change-Id: I84c1912867e2794502e8a63302c938a0cbab2c4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169545
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30661}
This is needed to be able to use webrtc::video_coding::EncodedFrame
is unit tests in Chromium.
TBR=tommi@webrtc.org
Bug: webrtc:11380
Change-Id: Idb3b0ab667a548f5a968e02a8efd91f02585c3f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169451
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30651}
~3-5% speed up on webrtc_perf_tests of vp9 on linux desktop.
Avoid going thru a lot of unnecessary code checks.
Change-Id: I2cb0d794bcf239c5057dfc04cd07a496f89a5016
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167640
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30586}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
This relands commit abf73de8ea.
with adjustments.
Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
It was removed from tests in https://webrtc-review.googlesource.com/c/src/+/123540.
If simulcast is not used, SimulcastRateAllocator returns the
same allocation as DefaultVideoBitrateAllocator.
Bug: webrtc:10164
Change-Id: I3d3e1aefe2fcc2bf853cd63c75e008b86eff9241
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168496
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30509}
This reverts commit abf73de8ea.
Reason for revert: breaks downstream tests
Original change's description:
> Do not propagate generic descriptor on receiving frame
>
> It was used only for the frame decryptor.
> Decryptor needs only raw representation that it can recreate
> in a way compatible with the new version of the descriptor.
>
> Bug: webrtc:10342
> Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30501}
TBR=danilchap@webrtc.org,sprang@webrtc.org,philipel@webrtc.org
Change-Id: I6634df06ee75aa8cdfda614994ab11f7a5845c70
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168488
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30502}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
Bug: webrtc:10342
Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30501}
If e.g. CPU adaptation reduces input video size too much, video pipeline would
reduce the number of used simulcast streams/spatial layers. This may result in
disabled video if some streams are disabled by Rtp encoding parameters API.
Bug: webrtc:11319
Change-Id: Id7f157255599dcb6f494129b83477cda4bea982a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168480
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30498}
The Resource interface (previously a skeleton not used outside of
testing) is updated to inform listeners of changes to resource
usage. Debugging methods are removed (Name, UsageUnitsOfMeasurements,
CurrentUsage). The interface is implemented by
OveruseFrameDetectorResourceAdaptationModule's inner classes
EncodeUsageResource and QualityScalerResource.
The new ResourceUsageListener interface is implemented by
OveruseFrameDetectorResourceAdaptationModule. In order to avoid adding
AdaptationObserverInterface::AdaptReason to the ResourceUsageListener
interface, the module figures out if the reason is "kCpu" or "kQuality"
by looking which Resource object triggered
OnResourceUsageStateMeasured(). These resources no longer need an
explicit reference to OveruseFrameDetectorResourceAdaptationModule and
could potentially be used by a different module.
In this CL, AdaptationObserverInterface::AdaptDown()'s return value is
still needed by QualityScaler. This is mirrored in the return value of
ResourceUsageListener::OnResourceUsageStateMeasured(). A TODO is added
to remove it and a comment explains how the current implementation
seems to break the contract of the method (as was the case prior to
this CL).
Follow-up work include:
- Move EncodeUsageResource and QualityScalerResource to separate files.
- Make resources injectable, allowing fake resources in testing and
removing OnResourceOveruseForTesting() methods.
(Investigate adding the necessary input signals to the Resource
interface or relevant sub-interfaces so that the module does not need
to know which Resource implementation is used.)
- And more! See whiteboard :)
Bug: webrtc:11222
Change-Id: I0a46ace4a2e617874e3ee97e67e3a199fef420a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30469}
Move definitions of mock classes to the only user, the unit tests for
the deprecated class vcm::VideoReceiver.
Bug: webrtc:7408
Change-Id: I05e38ed8ebbe615bb2db0b631ec914773fb0a520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168182
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30451}
number of references can't be invalid if gof was correctly parsed
from a vp9 packet, but RtpFrameReferenceFinder still better be
protected from the invalid data.
Bug: chromium:1048013
Change-Id: I548f5c87199421b7736409cbcacbec760ad799ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168124
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30444}
To make it generally faster, specially in case of very large picture id gaps.
Bug: None
Change-Id: Ib0c49c17bd1281190da986def43bea8fc3440c0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168055
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30438}
This CL uses |width| and |height| in RTPVideoHeaderVP9 to pass information
about enabled layers from encoder to packetizer.
Bug: webrtc:11319
Change-Id: Idc1c337f8dfb3f7631506acb784d2a634b41b955
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167724
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30428}
wrap ids before unwrapping: should be noop for ids arrived from the
network, but avoids DCHECKs for ids arrived from fuzzer.
for vp9 double check number of references doesn't exceed maximum.
for vp8 drop key frames for non-zero temporal id.
for general by seqnum code path do not set last_picture_id_:
it is not used there, but may confuse vp8 codepath.
as a slight speed up avoid copying RTPVideoTypeHeader for vp8 and vp9.
Bug: chromium:1046995, chromium:1047024, chromium:1047095, chromium:1047165, chromium:1047190
Change-Id: I1ab0833d32e2c023cbf5e3cfcc9e74f1c558e44b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168040
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30426}
Also updated FrameBuffer unittests to use the GlobalSimulatedTimeController.
Bug: webrtc:7408, webrtc:9378
Change-Id: I8ade27492f66cdd8950b38f5f4a268714dbc35fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164536
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30422}
Chromting is trying vp9 444 to have better color. This fix is needed to decode 444 properly.
Bug: webrtc:11326
Change-Id: I4498930591d8876af9f6b7238a8c9fe450ecbfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166220
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30410}
To allow to use the RtpFrameReferenceFinder with
an updated version of the frame descriptor extension
Bug: webrtc:10342
Change-Id: Ib60a505a714993862a008300aa64d0bb835c3377
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167361
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30407}
This allows for the possiblity to move the QualityScaler
out of the VideoStreamEncoder in the future.
Bug: webrtc:11222
Change-Id: I1d563cf08791e27ff5065ce90bcb150a7974d868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167534
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30406}
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.
Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
To free up RtpVideoHeader::generic field for codec agnostic details
from an rtp header extension.
Bug: webrtc:10342
Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30396}
