The overshoot detector uses a simple pacer model to determine an
estimate of how much the encoder is overusing the target bitrate.
This utilization factor can then be adjuster for when configuring the
actual target bitrate.
Spatial layers (simulcast streams) are adjusted separately.
Temporal layers are measured separately, but are combined into a single
utilization factor per spatial layer.
Bug: webrtc:10155
Change-Id: I8ea58dc6c4871e880553d7c22202f11cb2feb216
Reviewed-on: https://webrtc-review.googlesource.com/c/114886
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26573}
It doesn't do anything any more, so it should be removed.
Bug: webrtc:9586
Change-Id: I0b320b6ce4f480ff8cb59451db29bcc77b882b5f
Reviewed-on: https://webrtc-review.googlesource.com/c/120813
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26507}
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}
Further, strictly require VideoReceiveStream::Config::rendererer
to be non-null when the VideoReceiveStream is started. This is
already true by construction in the production code.
Bug: None
Change-Id: Ia0a41cfafa44215efc195a9eb6204194930c3dde
Reviewed-on: https://webrtc-review.googlesource.com/c/115040
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26384}
kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't
really beling on VideoBitrateAllocation.
common_types.h is going away and it feels dubious to requrie include
of the full VideoEncoder api to use them. Therefore moving them into a
seprate file/target.
Also includes some remaining cleanup of includes.
Bug: webrtc:9271
Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697
Reviewed-on: https://webrtc-review.googlesource.com/c/117305
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26299}
Preparation for replacing use of TestVideoCapturer as an interface,
instead using VideoSourceInterface.
Methods kept as non-virtual on the subclass FrameGeneratorCapturer,
but it's changed to be started on creation.
Bug: webrtc:6353
Change-Id: Iae1c9a0ee55d730d4992204f62227ef2f057d58e
Reviewed-on: https://webrtc-review.googlesource.com/c/114425
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26037}
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}
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}
All implementations of ModifyReceiverCallConfig and
ModifySenderCallConfig configure the bitrate_config member only. So
replace these methods by ModifyReceiverBitrateConfig and
ModifySenderBitrateConfig.
This is a preparation for injecting RtpTransportControllerSend via
CallConfig. Then bitrates should be passed when constructing
RtpTransportControllerSend, and they can be deleted from CallConfig.
Bug: webrtc:7135
Change-Id: I6714158bd463dd485018713d0e26815919e5afcc
Reviewed-on: https://webrtc-review.googlesource.com/c/110780
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25624}
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}
There was a suggestion in a previous CL to add an end to end test case to
prevent future regressions. I have enabled this by adding two fakes that
perform fake encryption and enabling an end to end test with VP8 and the
GenericDescriptor.
Bug: webrtc:9927
Change-Id: Icf96eeed541ada1e0579eb81b6f87a46d1c43d96
Reviewed-on: https://webrtc-review.googlesource.com/c/108020
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25399}
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}
Use in VideoQualityTest replaced by creating a wrapper for the decoder,
similarly to https://webrtc-review.googlesource.com/94152 which
deleted the corresponding method on VideoSendStream.
Bug: webrtc:9106
Change-Id: I0a7798bc44704af8b36017655b9ffa34fa1423e6
Reviewed-on: https://webrtc-review.googlesource.com/97580
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24926}
This is a reland of 529d0d9795b81dbed5e4231f15d3752a5fc0df32
Original change's description:
> Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
>
> Preparation for deleting EnableFrameRecordning, and also a step
> towards landing of the new VideoStreamDecoder.
>
> Bug: webrtc:9106
> Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
> Reviewed-on: https://webrtc-review.googlesource.com/97660
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24861}
Bug: webrtc:9106
Change-Id: I2eb894773b3f33ff6a980e8008e8248607e32668
Reviewed-on: https://webrtc-review.googlesource.com/102480
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24882}
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}
Preparation for deleting EnableFrameRecordning, and also a step
towards landing of the new VideoStreamDecoder.
Bug: webrtc:9106
Change-Id: I50964ee458b08a702ec69b82a62e4995c57cee82
Reviewed-on: https://webrtc-review.googlesource.com/97660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24861}
This reverts commit 3f4a4fad8cd661309ff5d9a631e89518f32e7c5e.
Reason for revert: Breaking internal tests
Original change's description:
> Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
>
> Also parameterized tests to test the new generic descriptor and
> added --generic_descriptor flag to loopback tests.
>
> Bug: webrtc:9361
> Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> Reviewed-on: https://webrtc-review.googlesource.com/101900
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24835}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/101940
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24839}
Also parameterized tests to test the new generic descriptor and
added --generic_descriptor flag to loopback tests.
Bug: webrtc:9361
Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
Reviewed-on: https://webrtc-review.googlesource.com/101900
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24835}
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}
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}
Use in VideoQualityTest replaced by creating a wrapper for the encoder.
Bug: None
Change-Id: I5c5519e147ca7ddb97696b0d6958a8a1f5cc6e83
Reviewed-on: https://webrtc-review.googlesource.com/94152
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24533}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
This is reland after fixes for broken perf tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/91861
Bug: none
Change-Id: I6999a499408787be43a74a26a16b7826a0814a7b
Reviewed-on: https://webrtc-review.googlesource.com/95182
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24383}
