Use speed 6 for better quality for low resolution, speed 8 for HD for better speed.
This will better balance speed and quality.
Change-Id: I3d8dbd45533471ce58d53c1ac26f92c7b1106259
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175281
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31336}
to convert flags which chains a video frame part of into chain_diffs
Bug: webrtc:10342
Change-Id: I6fb899eae934078223b101c9f85e2ac101980d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175108
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31306}
Add TODOs into AV1 encoder wrapper where it suppose to be used.
Bug: webrtc:11404
Change-Id: If049066b84be72829867d5084827a7d275648a7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174806
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31278}
Max encoder bitrate in WebRTC and OpenH264 are different settings. In
WebRTC it is a cap for encoder target bitrate whilst in OpenH264 it is
a peak bitrate. I.e. OpenH264 is allowed to produce bitrate up to
iMaxBitrate for short time interval. That is not what WebRTC expects.
https://webrtc.googlesource.com/src/+/5ee6967c4edc667688d736c27db6f2e7be00dd0a
disabled encoders re-initialization on min/max bitrate change. Reinit of
some HW encoders takes hundreds of milliseconds and causes video freeze.
I missed that max bitrate is used by OpenH264. This caused regression
described in webrtc:11543.
This change sets iMaxBitrate=UNSPECIFIED_BIT_RATE (which is the default
value). Settings iMaxBitrate=UNSPECIFIED_BIT_RATE disables the frame
dropping logic based on that parameter. But the encoder still will drop
frames based on buffer fullness, https://source.chromium.org/chromium/chromium/src/+/master:third_party/openh264/src/codec/encoder/core/src/ratectl.cpp;l=806-807
Bug: webrtc:10773, webrtc:11543
Change-Id: I728be49e0df8a0d9a8f4438299e4c7b4c1497a78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174745
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31192}
while helpful by itself, it is also a preparation
for adding unittests for (to be added) svc features of the encoder.
Bug: webrtc:11404
Change-Id: I62b0645f44579f21f228d406a206b4c01d80dd02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174580
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31189}
Call is instantiated on what we traditionally call the 'worker thread'
in PeerConnection terms. Call statistics are however gathered, processed
and reported in a number of different ways, which results in a lot of
locking, which is also unpredictable due to the those actions themselves
contending with other parts of the system.
Designating the worker thread as the general owner of the stats, helps
us keeps things regular and avoids loading unrelated task queues/threads
with reporting things like histograms or locking up due to a call to
GetStats().
This is a reland of remaining changes from https://webrtc-review.googlesource.com/c/src/+/172847:
This applies the changes from the above CL to the forked files and
switches call.cc over to using the forked implementation.
Bug: webrtc:11489
Change-Id: I93ad560500806ddd0e6df1448b1bcf5a1aae7583
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31186}
Inserting old frames is not an error condition and should not print a warning, and given that it happens all the time it is also very spammy.
Bug: chromium:1066819
Change-Id: Iad2b5edc5e62822c02e2bb2a53d4318f957be3bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173022
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31172}
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:
- RTC_OBJC_TYPE_PREFIX:
Macro used to prepend a prefix to the API types that are exported with
RTC_OBJC_EXPORT.
Clients can patch the definition of this macro locally and build
WebRTC.framework with their own prefix in case symbol clashing is a
problem.
This macro must only be defined by changing the value in
sdk/objc/base/RTCMacros.h and not on via compiler flag to ensure
it has a unique value.
- RCT_OBJC_TYPE:
Macro used internally to reference API types. Declaring an API type
without using this macro will not include the declared type in the
set of types that will be affected by the configurable
RTC_OBJC_TYPE_PREFIX.
Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10
The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.
Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).
QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.
The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.
Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31140}
This is more logical way to remove inactive lower layers.
Current way is to notify the encoder that the layer is inactive,
then renumber layers at the packatization level.
This Cl will allow to simplify libvpx vp9 encoder, svcRateAllocator and
vp9 packetizer.
Bug: webrtc:11319
Change-Id: Idf0bb30b729f5ecc97e31454b32934546b681aa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173182
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31058}
This reverts commit 8e8b36a94a.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
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}
