This reverts commit 6958d2c6f0.
Disable the test on iOS.
Bug: None
Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31437}
to unblock rolling new version where private function is no longer available
Bug: None
Change-Id: I9c35fede3f331f7688cc97acfbda1250b42348a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176441
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31427}
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}
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}
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 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}
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}
~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}
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}
