This is a reland of 32ca95145c4636374266f5b5d4d1ac43658bc758
Fix includes not enabling the screenshare conference behavior on non
screenshare sources even if the flag is enabled.
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
Bug: webrtc:11310
Bug: chromium:1093819
Change-Id: Ic933f93a5c4bad20583354fe821f8a1170e911cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180802
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31847}
This reverts commit 32ca95145c4636374266f5b5d4d1ac43658bc758.
Reason for revert: Internal test failure
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
TBR=ilnik@webrtc.org,hta@webrtc.org,orphis@webrtc.org
Change-Id: I5ccb6e87594f491ba09fe6b837ee24d63db878ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11310
Bug: chromium:1093819
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180801
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31829}
Non-conference mode simulcast screenshares were mistakenly using the
conference mode semantics in the simulcast rate allocator, which broke
spec compliant usage in some situation.
This behavior should only be used when explicitly using the SDP entry
"a=x-google-flag:conference" in both offer and answer.
Bug: webrtc:11310, chromium:1093819
Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31828}
RTPFragmentationHeader is already ignored by H264 packetizer
and thus doesn't need to be provided and calculated.
Bug: webrtc:6471
Change-Id: I45bc22827f0dc811457e3ebe477a16293501c2fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179843
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31791}
This is one less dependency on the task queue, and will make
things like removing resources and cleanup much easier in the future.
Bug: webrtc:11754
Change-Id: I732f1935d1b58ffe09ca2a2bf59beebc1930214d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178869
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31686}
This is a step needed for multi-stream and new mitigations. It also
cleans up needing to signal adaptation changes in mutiple places
from ResourceAdaptationProcessor.
R=hbos@webrtc.org
Bug: webrtc:11754
Change-Id: Ib185dc9f66fbb4a087eb9e970c68c3f47eafb17f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178874
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31684}
put default values next to the (named) members.
Remove constructor
declared as VideoQualityTestFixtureInterface::Param::Param,
but defined as VideoQualityTest::Param::Param
avoid braced initialisation for structures with lot's of fields that change over time
Bug: None
Change-Id: I647e3ace47ae981c515d6939274cb2744dd1b866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178811
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31674}
Changes in effective degradation preference is now done directly
on the VideoStreamAdapter. This means that the
DegradationPreferenceListener class can be removed.
Bug: webrtc:11700
Change-Id: I9ce4b627de0279df5ef2e5a38435a6d352fbb135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178609
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31642}
It has no relevance in ResourceAdaptationProcessorInterface.
The logic moves to an interaction directly on the VideoStreamAdapter.
Bug: webrtc:11700
Change-Id: I4c7b3e1e1361722a69b71e8f9bde33f5909c011a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178380
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31634}
This patch creates a new class which provides the DegradationPreference
thread safe to all classes that need if (BalancedConstraint and
QpScalerResource). It also broadcasts to all listeners when the
preferences are updated, so the ResourceAdaptationProcessor can update
the video if needed.
In future work, this could be used to remove the need for two task
queues for the VideoStreamEncoder resources.
Bug: webrtc:11700
Change-Id: I05480db8b7157b5643f6f86ec9c64850839b3e76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177522
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31623}
This moves this responsibility from the ResourceAdaptaitonProcessor
to the VideoStreamAdapter. A new interface for listening to adaptation
changes was added, and the ResourceAdaptationProcessor now listens on
the VideoStreamAdapter for those changes.
This means that going forward,
1. We can do adaptations outside of resource limitations, like setting
prior adaptations on a resource like initial frame dropper is designed
to.
2. Adaptations can be on a different queue than the
ResourceAdaptaitonProcessor's queue since updates are pushed through
the listener.
Bug: webrtc:11700
Change-Id: I6de0dec748dba095e702f0b9893c5afa50b51aa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176859
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31615}
Add TimeController to the CreatePeerConnectionE2EQualityTestFixture
method as a first step to make PC level framework compatible with
TimeController abstraction.
Bug: webrtc:11743
Change-Id: I69305abc880059bf9fe1d4f2e3b7c10cf35417db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178485
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31607}
This reduces the number of times we grab a few locks down from
somewhere upwards of around a thousand time a second to a few times.
* Update the RTT value on the worker thread and fire callbacks.
* Trigger NotifyTmmbrUpdated() calls from the worker.
* Update the tests to use a GlobalSimulatedTimeController.
Change-Id: Ib81582494066b9460ae0aa84271f32311f30fbce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177664
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31602}
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.
Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
1) Fix several typos and small mistakes which could lead to crashes
2) Adjust bitrates if leading layers are disabled
3) Wire up webrtc quality scaler
Bug: webrtc:11319
Change-Id: I16e52bdb1c315d64906288e4f2be55fe698d5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177525
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31546}
When a resource is removed from the ResourceAdaptationProcessor,
and the resource is the most limited resource, we reset the current
adaptation to the next most limited resource level. In the case that
this resource is the only adapted resource then we reset all adaptations.
Bug: webrtc:11636
Change-Id: I29acc5a3934f42f00db79b3bb2171156b1313937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176406
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31520}
This CL adds AddAdaptationResource to Call and
AddAdaptationResource/GetAdaptationResources method to relevant
VideoSendStream and VideoStreamEncoder interfaces and implementations.
Unittests are added to ensure that resources can be added to the Call
both before and after the creation of a VideoSendStream and that the
resources always gets added to the streams.
In a follow-up CL, we will continue to plumb the resources all the way
to PeerConnectionInterface, and an integration test will then be added
to ensure that injected resources are capable of triggering adaptation.
Bug: webrtc:11525
Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31499}
A more detailed explaination is in the bug, but this changes
the way that adaptation happens when multiple resources are
limited. Only the one that is most limited can trigger an
adaptation up. If multiple resources are most limited both
need to underuse to adapt up.
Some of the changes in this patch to make it all work:
* VideoStreamEncoder unittests that did not reflect this
new behaviour have been changed.
* PeekNextRestrictions returns the adaptation counters as
well as the restrictions.
* Adaptation statstics have changed so that when adapting
up all resources are tagged as triggering the adaptation.
Additionally the statistics for the current adaptation is
now the total number of adaptations per reason, rather then
the number of adaptations due to that reason.
* PreventAdaptUpDueToActiveCounts is removed as most limited
resource is a strong implementation of that.
Bug: webrtc:11553
Change-Id: If1545a201c8e019598edf82657a1befde8b05268
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176128
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31497}
Unit test added to verify root cause is fixed.
Scenario test added to verify high-level behavior.
Bug: webrtc:11654
Change-Id: I1ad6e2750f5272e86b4198749edbbf5dfd8315c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176564
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31462}
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.
Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.
The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.
Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
The method is being used externally to create instances
of the deprecated internal implementation.
Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.
Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}
