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}
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}
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}
IsAdaptationUpAllowed is moved from Resource to AdaptationConstraint.
OnAdaptationApplied is moved from Resource to AdaptationListener.
In a future CL, Resource will be moved to api/, but
AdaptationConstraint and AdaptationListener will stay in call/.
The processor, encode stream and manager are updated to keep track of
both resources, constraints and listeners. Fakes and tests are updated.
After this CL, the manager's inner classes that prevent adaptation
implement AdaptationConstraint instead of Resource.
Bug: webrtc:11525
Change-Id: Ie9cd5b1ba7d8e161951e131ab8f6bd9d5cf765bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176368
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31409}
This CL is in preparation for moving Resource to the api/ folder. It
does not move it, but makes it such that the moving CL can be a pure
move.
In order to do this, we must stop depending on rtc_base/rtc::TaskQueue
in favor of api/webrtc::TaskQueueBase.
There are also other rtc_base/ dependencies that we do not want to
expose to the api/ folder, like critical sections and thread
annotations which are not publically exposed. To get around this, we
make Resource an abstract interface and move all of the base class
functionality into a new non-api/ class: VideoStreamEncoderResource.
The Resource now has Register/UnregisterAdaptationTaskQueue() methods.
By explicitly unregistering, we can ensure validity of the pointer even
if the Resource outlives the PeerConnection. While public interface
methods are only to be called on the adaptation task queue, posting to
the task queue happens off-queue, so a |lock_| is introduced to guard
it.
Bug: webrtc:11525
Change-Id: I50b3a30960cdec9032016c779b47001c01dad32f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176320
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@{#31402}
This only replaces the methods with the signatures VerifyX(wants,
value). This ensures that failing expectations give the corrent line,
where previously it gave the line in the helper methods.
Bug: None
Change-Id: I89e883ccd15b3ea5ce364d2c16b0c3ac219f139c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175912
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31345}
Some VideoStreamEncoderTests such as
AdaptsFramerateForLowQuality_MaintainResolutionMode had been observed
to be flaky. In a local run, this test failed 1/12000 times, but on
bots it may have failed more often.
All tests could sometimes fail due to expecting something to be the
case without waiting for that thing to happen, which was made evident
when adding SleepMs() at strategic points in the code and running the
tests locally.
With this CL, the tests should no longer be flaky after having added
waiting for adaptation to be applied or EXPECT_TRUE_WAIT for sink wants
to have the appropriate values.
Bug: webrtc:11586
Change-Id: I60e76c742d9ccc8305feb14a834a4f61a60a62a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175654
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31327}
This test ensures that when changing degradation preference,
a resource that was previously downgraded in a different degradation
preference can not adapt up.
Bug: webrtc:11522, webrtc:11523
Change-Id: Id362530408db4c49b0d0b2516be9a11ccc7c8f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175012
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31250}
(Misc cleanup associated with
https://webrtc-review.googlesource.com/c/src/+/174441.)
This test was previously assuming what when QP usage is handled it
first posts to the adaptation queue and then back with the result from
the encoder queue.
While the assumption is correct it is not an implementation detail that
the test was trying to assert, nor do we need such a test.
TriggerQualityScalerHighQpAndReturnIfQpSamplesShouldBeCleared() is
updated to wait for an event associated with QP having been handled,
which is all that the test really cares about.
Bug: webrtc:11542, webrtc:11520
Change-Id: I3286c3ab631f09c43abe0fd59f31c3997aedd9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175004
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31243}
This CL unblocks future Call-Level Mitigation strategies by moving the
ResourceAdaptationProcessor to a separate task queue. This signifies a
major milestone in the new resource adaptation architecture because
with this CL the threading model is in place and moving the Processor
to the Call and increasing its responsibilities is made possible.
In this CL, we still have one Processor per VideoStreamEncoder and the
VideoStreamEncoder is responsible for the creation and the destruction
of its Processor and that Processor's task queue. But the PostTasks are
in place and the decision-making is executed on a separate queue.
This CL:
- Moves ResourceAdaptationProcessor to an adaptation task queue.
It continues to be entirely single-threaded, but now operates on a
separate task queue.
- Makes Resources thread-safe: Interaction with the Processor, i.e.
OnResourceUsageStateMeasured() and IsAdaptationUpAllowed(), happens
on the adaptation task queue. State updates are pushed from the
encoder task queue with PostTasks.
- QualityScalerResource operates on both task queues; the QP usage
callbacks are invoked asynchronously.
- The VideoStreamEncoderResourceManager operates on the encoder task
queue with the following exceptions:
1) Its resources are accessible on any thread (using a mutex). This
is OK because resources are reference counted and thread safe.
This aids adding and removing resources to the Processor on the
adaptation task queue.
2) |active_counts_| is moved to the adaptation task queue. This makes
it possible for PreventAdaptUpDueToActiveCounts to run
IsAdaptationUpAllowed() on the adaptation task queue.
A side-effect of this is that some stats reporting now happen on
the adaptation task queue, but that is OK because
VideoStreamEncoderObserver is thread-safe.
The Manager is updated to take the new threading model into account:
- OnFrameDroppedDueToSize() posts to the adaptation task queue to
invoke the Processor.
- OnVideoSourceRestrictionsUpdated(), now invoked on the adaptation
task queue, updates |active_counts_| synchronously but posts to the
encoder task queue to update video source restrictions (which it
only uses to calculate target frame rate).
- MaybePerformQualityRampupExperiment() posts to the adaptation task
queue to maybe reset video source restrictions on the Processor.
|quality_rampup_done_| is made std::atomic.
Bug: webrtc:11542, webrtc:11520
Change-Id: I1cfd76e0cd42f006a6d2527f5aa2aeb5266ba6d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174441
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@{#31231}
In a future CL, adaptation processing and stream encoder resource
management will happen on different task queues. When this is the case,
asynchronous tasks will be posted in both directions and some resources
will have internal states used on multiple threads.
This CL makes the Resource class reference counted in order to support
posting tasks to a different threads without risk of use-after-free
when a posted task is executed with a delay. This is preferred over
WeakPtr strategies because WeakPtrs are single-threaded and preferred
over raw pointer usage because the reference counted approach enables
more compile-time and run-time assurance. This is also "future proof";
when resources can be injected through public APIs, ownership needs to
be shared between libwebrtc and the application (e.g. Chrome).
To reduce the risk of making mistakes in the future CL, sequence
checkers and task queue DCHECKs are added as well as other DCHECKs to
make sure things have been cleaned up before destruction, e.g:
- Processor gets a sequence checker. It is entirely single-threaded.
- Processor must not have any attached listeners or resources on
destruction.
- Resources must not have any listeners on destruction.
- The Manager, EncodeUsageResource and QualityScalerResource DCHECKs
they are running on the encoder queue.
- TODOs are added illustrating where we want to add PostTasks in the
future CL.
Lastly, upon VideoStreamEncoder::Stop() we delete the
ResourceAdaptationProcessor. Because the Processor is already used in
posted tasks, some if statements are added to ensure the Processor is
not used after destruction.
Bug: webrtc:11542, webrtc:11520
Change-Id: Ibaa8a61d86d87a71f477d1075a117c28d9d2d285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174760
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31217}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
The ResourceListenerResponse was used to make the QualityScaler
not clear QP samples and instead increase its frequency of checking for
QP under certain circumstances, see enum description:
https://webrtc.googlesource.com/src.git/+/c70b1028d47c1aee4892545190cd66e97d09cd55/call/adaptation/resource.h#33
Because the QualityScaler depends on whether and how adaptation
happened it should listen to adaptation happening.
This CL moves the logic that was previously in VideoStreamAdapter closer
to the QualityScaler: QualityScalerResource::OnAdaptationApplied().
This would allow the VideoStreamAdapter to operate on a separate task
queue in the future, with no dependencies on any stream-specific
resources that might operate on other task queues.
Bug: webrtc:11172, webrtc:11521
Change-Id: I07971a8a5fab5715f4ccb7d2c63f1b92bd47170f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173090
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31143}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
The ResourceUsageState was written as: {kOveruse, kStable, kUnderuse}.
The assumption was that if a resource neither wanted to adapt up or
down it would report kStable. But with the addition of
Resource::IsAdaptationUpAllowed() (prior CL) the notion of being
"stable" was already captured outside of ResourceUsageState.
Furthermore, kStable failed to capture what IsAdaptationUpAllowed() did
not: whether we can go up depends on the resulting resolution or frame
rate (restrictions_after). Perhaps we can go up a little, but not a lot.
This CL also adds Resource::ClearUsageState(). After applying an
adaptation, all usage states become invalidated (new measurements are
needed to know if we are still over- or underusing). This was always
the case, but prior to this CL this was not accurately reflected in the
Resource::usage_state() in-between measurements.
Bug: webrtc:11172
Change-Id: I140ff3114025b7732e530564690783e168d2509b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173088
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@{#31110}
Replaces this with 2 methods instead, adding clarity.
ClearAdaptationStats
- Resets the adaptations statistics to 0. This is done,
when the degredation is reset, for example when the preference
is changed to/from BALANCED.
UpdateAdaptationMaskingSettings
- Updates the settings for adaptation statistics reporting.
This way we don't report quality adaptations if quality scaling
is not enabled (same for resolution/fps scaling).
The adaptation counting inside the SendStatisticsProxy is
now done in a struct that counts the totals, and then masks
out these counts based on the adaptation settings. The
MaskedAdaptationSteps uses optionals to hide the values we
shoudn't report, while the AdaptationSteps always hold the real
totals.
All tests have been updated to use the Reset/Clear method as needed.
Now that AdaptationCounters and AdaptSteps use the same structure,
AdaptationCounters was moved to api/video and replaces AdaptSteps.
The AdaptReason enum is also redundant now, and will be removed
in a follow-up CL.
R=hbos@webrtc.org
Bug: webrtc:11392
Change-Id: Iaed6488581325d341a056b5bbf76a01c19d6c282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171685
Reviewed-by: Niels Moller <nisse@webrtc.org>
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@{#31083}
The MockFecController is spitting out lots of warnings, as it is
being called when we don't care about it, in normal tests. Making
it a NiceMock allows it to receive calls without expectation and
not warn.
Bug: None
Change-Id: I1ea219c4665d86917718692dc013ae3ac47222ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30820}
There is still a counter for the active counts for the
scaling, but these will be removed at a later date.
BUG=webrtc:11392
Change-Id: Ie9bcf3f744a0bbac601f0da61197f4bac1e9f879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169447
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30701}
This CL takes us one step closer to being able to evaluate alternative
possible adaptation targets (e.g. multi-stream adaptation) by exposing
the target separately from applying it.
This is a refactoring of OnResourceUnderuse() and OnResourceOveruse().
Prior to this CL, the target resolution or frame rate was calculated
inside these methods and applied if possible. This CLs makes these two
steps (calculating a usable target + applying it) separate methods.
After this CL, the target is expressed as AdaptationTarget and is
calculated and returned by GetAdaptUpTarget() and GetAdaptDownTarget().
The target is only returned if it can be applied - and CanAdaptUp() +
CanAdaptDown() are merged with these methods.
Applying the target happens at ApplyAdaptationTarget().
Bug: webrtc:11222
Change-Id: I8e488be1d1590c23848db816d49a7738562e176d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30643}
* This replaces the video stream methods for forcing adaptation
with a mock resource that triggers overuse.
* Resources can now be injected to the Module using the AddResource
function.
* Resources now have tests for adding and removing callbacks.
* Quality/EncoderUse% resources are tracked in the Resource list of
the adaptation module.
* The adaptation module ties all resources to a reason to keep stats
working as expected.
BUG=webrtc:11377
Change-Id: I1f5902f7416dc41b4915c0072e6f0da2bb3bb2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168948
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@{#30610}
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}
The EncoderSelectorInterface is meant to replace the "WebRTC-NetworkCondition-EncoderSwitch" field trial, so the field trial will be ignored if an EncoderSelectorInterface object has been injected.
Bug: webrtc:11341
Change-Id: I5371fac9c9ad8e38223a81dd1e7bfefb2bb458cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168193
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30490}
With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.
Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
This refactors the usage of OveruseFrameDetector in
OveruseFrameDetectorResourceAdaptationModule into an inner class of the
module, making the interaction between the detector and the module the
responsibility of this helper class instead.
Similarly, QualityScaler usage is moved into QualityScalerResource.
This takes us one step closer to separate the act of detecting
overuse/underuse of a resource and the logic of what to do when
overuse/underuse happens.
Follow-up CLs should build on this in order to materialize the concept
of having resources, streams and a central decision-maker deciding how
to reconfigure the streams based on resource usage state.
Bug: webrtc:11222
Change-Id: I99a08a42218a871db8f477f31447a6379433ad05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168057
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30468}
Before this change, if both app and encoder provided bitrate limits,
WebRTC ignored the limits provided by encoder. Now intersection of
these sets is used.
Also changed DCHECKs in GetEncoderBitrateLimits to allow zero values
of min_bitrate_bps and min_start_bitrate_bps.
Bug: none
Change-Id: Ib8be965ea43f51013b0a0f82fd4256a372432dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166600
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30338}
This moves SetHasInputVideoAndDegradationPreference() to the encoder
queue. OveruseFrameDetectorResourceAdaptationModule is now entirely
single-threaded, including its inner class VideoSourceRestrictor.
VideoStreamEncoder now protects the module with RTC_GUARDED_BY. This
ensures it is safely used, even without a SequenceChecker inside of the
module. The module's |encoder_queue_| is removed.
The one task queue reference that is needed - passing down the current
task queue to StartCheckForOveruse() - is replaced by a TaskQueueBase*
(instead of rtc::TaskQueue*), enabling obtaining the current queue with
TaskQueueBase::Current(). (There is no rtc::TaskQueue::Current().)
Furthermore, the only uses of VideoSourceSinkController that isn't on
the encoder queue are documented, with a TODO saying if these are moved
the VideoSourceSinkController could also be made single-threaded.
However since this requires introducing a delay to
VideoStreamEncoder::SetSource() and VideoStreamEncoder::Stop(),
arguably a more risky change, if this is to be attempted that should be
in a separate CL.
Bug: webrtc:11222
Change-Id: I448ca5125708d5f66b95b0b180d6d24cc356dfa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165783
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30263}
This CL puts the VideoStreamEncoder's current adaptation logic inside
the new class OveruseFrameDetectorResourceAdaptationModule. The
intention is not to change any behavior, only to move code.
Future CLs should step by step decrease the coupling between
OveruseFrameDetectorResourceAdaptationModule, VideoStreamEncoder and
the VideoStreamEncoder's QualityScaler by introducing more abstract
interfaces. This is not done in this CL because it is large enough as
it is, but the long term goal is to make it possible to replace the
existing overuse module with a different implementation.
This CL relies on existing tests exercising the VideoStreamEncoder, but
part of making overuse logic modular should include testing each module
separately as well as continued integration testing of the
VideoStreamEncoder.
Bug: webrtc:11222
Change-Id: I316a174adfd00d60cdd224a23a5f616efd235d13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161953
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30163}
If WebRTC-AutomaticAnimationDetectionScreenshare experiment is enabled,
content type is screenshare and degradation preference is BALANCED,
then input resolution is restricted if update_rect of the incoming frames
is the same for considerable amount of time and is big enough.
This entails treating BALANCED degradation preference for screenshare as
MAINTAIN_RESOLUTION in adaptation logic.
Bug: webrtc:11058
Change-Id: I903dddf53fcbd7c8eac6c5b1447225b15fd8fe5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161097
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30002}
This change makes it easier to propagate more information from the sink
to the video adapter, for example alignment requirements.
Bug: None
Change-Id: I536248d59f871c103a18a48615b6c5e61f61697b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29827}
- Use minimum start bitrate to drop frame and adapt resolution in the
beginning of call.
- Use minimum bitrate to decide whether or not resolution should be
increased based on quality in MAINTAIN_FRAMERATE and BALANCED modes.
In BALANCED mode bitrate limits provided by the corresponding field
trial are prioritized over the limits provided by encoder.
Bug: webrtc:10853
Change-Id: I8257eb64565bcafa6ae9887a1af18e90f8400cac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156302
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29461}
This is a reland of 809198edff
A fix was made in https://webrtc-review.googlesource.com/c/src/+/154343
which fixed the regression issues caused by the original patch.
Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}
Bug: webrtc:10126
Change-Id: Iecc3ab6a5cd1193a1fa8e824dcf4f0b8165f9bf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154359
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29356}
Without invalidation the codec configuration may not get updated. This
is prone to happen when the bandwidth is below the min or above the max
in which case the last_encoder_rate_settings_ may have not changed.
This led to a regression in b6a45dd, where last_encoder_rate_settings_
would not change the allocation or frame rate, but the frame rate would
have been set by the video adapter. Thus the frame rates were set
incorrectly, leading to lower values in the regression tests.
I have re-run this scenario against some of the metric drops and the
regression appears to be fixed.
Bug: webrtc:10126
Change-Id: I0fa6c9b71e7aff5dd80e53119db109d97eed98b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154343
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29301}
This reverts commit 809198edff.
Reason for revert: Performance regressions that need to be addressed.
Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}
TBR=sprang@webrtc.org,eshr@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10126
Change-Id: I133cbe5d8cb894ed944ae8a2d0f63a78bbed72ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153484
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29221}
VideoEncoder::SetRates was being called unnessesarily when the fields
appended to RateControlParameters were changed. Since SetRates only
cares about RateControlParameters, it should have only been called if
the RateControlParameters themselves were actually changed.
Bug: webrtc:10126
Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#29208}
In this CL:
- Renamed EncoderFailureCallback to EncoderSwitchRequestCallback. An encoder
switch request can now also be made with a configuration that specifies which
codec/implementation to switch to.
- Added "WebRTC-NetworkCondition-EncoderSwitch" field trial that specifies
switching conditions and desired codec to switch to.
- Added checks to trigger the switch based on these conditions.
Bug: webrtc:10795
Change-Id: I9d3a9a39a7c4827915a40bdceed10b581d70b90a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29196}
The new target, modules/video_coding:video_coding_legacy, is not
depended upon by any webrtc non-test code.
Bug: webrtc:7408
Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29133}
The stable target rate is used to make smarter choices in the rate
to chose which layers to enable in SVC or simulcast modes.
the addition of hysteresis, we can improve a call quality by reducing
the amount of resolution switch.
Bug: webrtc:10126
Change-Id: I04d0df9e6bbe247e2f2a668207ff74d475e2464c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150642
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29112}
The VideoBitrateAllocator subclasses may actually allocate more than the
target, in order to satisfy the min bitrate constraint. In this case,
make sure the bandwidth allocation we signal to the encoder is at least
this large.
Bug: chromium:995462
Change-Id: I08b89a7c54392330d773e13c1b0a3eff42f81672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151125
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29040}
The ScopedFakeClock contains a lock. Due to declaration order, this is
the first member of VideoStreamEncoderTest to be destroyed. However,
there are cyclic tasks that may still be running at that time, and they
may try to read the time, so if we're unlucky they may trigger a use
after free condition.
This only affects test and is simply solved by moving the declaration
to before the classes that uses it.
Bug: webrtc:10929
Change-Id: I998d5ced877f355e4a45ee5cf75b2eb75faa6113
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150795
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29008}
Add separate setting for configuring min bitrate that only applies when
adapting up in resolution.
Bug: none
Change-Id: I83d33ac3110a22602065b8d83130e3f619cb1eba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150329
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28970}
If a framerate reduction (input fps - restricted fps) is less than the
configured diff, shorten interval to next qp check.
Bug: none
Change-Id: Ia0b9e0638e5ba75cdc20a1bb45bfcb7d858c5f89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149040
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28880}
Add possibility to configure min bitrate based on resolution.
Only adapt up if bw estimate is above the min bitrate for next higher resolution.
Bug: none
Change-Id: Ie38faae07d23336675ec33697ace6f6fed322efa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148598
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28863}
Also:
- DCHECK that bitrate limits recommended by encoder are reasonable.
- Restrict target bitrate such that it never exceed the max bitrate.
Bug: webrtc:10853
Change-Id: Ie43d30a7acfc8fa115deffd94165844248ce7945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148442
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28822}
optional<int> initial_bitrate_interval_ms: time interval since start of call
where fast adapt down is allowed.
optional<double> initial_bitrate_factor: try fast adapt down if bw estimate is
below initial bitrate * factor.
Bug: none
Change-Id: I63e1fdaac6556d8e9a961a42e11c925f9ecb9771
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147725
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28753}
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.
Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
If VideoEncoderConfig::max_bitrate_bps is unset then max bitrate of
video stream is set equal to max bitrate value recommended by encoder
for given resolution via encoder capabilities (if available).
Bug: webrtc:10796
Change-Id: I7fce9afc476b794a16956e694e891faee110048e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144526
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28515}
Before this change, an attempt to recreate video encoder would fail if
video encoder factory supports only single instance of an encoder.
Added tracking of max number of existed simultaneously encoder
instances to VideoEncoderProxyFactory.
Bug: webrtc:10776
Change-Id: I317cbdf1af94dfb4c72bf99c5cd4ce7b454188fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144044
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28457}
- Don't reset encoder if max/min bitrate changed.
- Removed min/max bitrate DCHECKs from encoder wrappers.
- Reset encoder if start_bitrate changed. Only do this if encoding
has not yet started.
- Updated ReconfigureBitratesSetsEncoderBitratesCorrectly test.
- Removed EncoderSetupPropagatesCommonEncoderConfigValues test since it
was a subset of ReconfigureBitratesSetsEncoderBitratesCorrectly.
Bug: webrtc:10773
Change-Id: Id9cbb2ea229232fd95967819e2a937b26948de9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144028
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28446}
Previously, FecControllerOverride was passed to
Vp8FrameBufferController::SetFecControllerOverride. Passing to
the factory is a more elegant way, since it's only used when
the controller is constructed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28443}
Some downstream clients have custom frame types that can't be converted.
The rest of EncodeVideoFrame is protected against these frames, but the
crop code assumes ToI420 always succeeds.
Bug: None
Change-Id: I8f4279e3975d3ae8cd1da59f7e84fafe0404fd15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141646
Commit-Queue: Noah Richards <noahric@chromium.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28256}
This is a reland of 11dfff0878
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
Reset() is called each time the encoder is reconfigured, but then it
happens the target bitrate isn't reset in encoder. So it might produce a
frame before next bitrate estimate is propagated to the metadata writer.
The incorrect zero bitrate would be treated as a paused encoder and would
cause metadata to be dropped.
Also, added unittest for that scenario at VideoStreamEncoder level.
Bug: webrtc:10460
Change-Id: I28024a527f1fb8474b172e2c5c2394fd38d69a07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140101
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28159}
This is a reland of 7ac0d5f348
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
This reverts commit 7ac0d5f348.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This CL wires up the new SetRates() method of the video encoders, and
refactors a few things in the process:
Most notably, the VideoStreamEncoderInterface is update so that the
|target_headroom| parameter is replaced with |link_allocation|, meaning
that instead of indicating bitrate capacity in excess of the target
bitrate, it indicates to total network capacity allocated for the
stream including the target bitrate. This matches the VideoEncoder API.
The VideoEncoder::RateControlParameters struct gets a few new helper
methods.
In VideoStreamEncoder, instead of adding more fields to the
|last_observed_bitrate*| family, uses an optional struct that
inherits from VideoEncoder::RateControlParameters.
Bug: webrtc:10481
Change-Id: Iee3965531142ae9b964ed86c0d51db59b1cdd61c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131123
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27487}
Fix issue where RTCP XR target bitrate could be incorrect when temporal layers are configured but it is not supported by encoder.
Bug: webrtc:10475
Change-Id: Ib525eb5f0ad8392e88d2579930ac8e459a1d194b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128778
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27268}
This CL paves the way to making FrameBufferController injectable.
LibvpxVp8Encoder can manage multiple streams. Prior to this CL,
each stream had its own frame buffer controller, all of them held
in a vector by LibvpxVp8Encoder. This complicated the code and
produced some code duplication (cf. SetupTemporalLayers).
This CL:
1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs
will make this factory injectable.)
2. Makes LibvpxVp8Encoder use a single controller. This single
controller will, in the case of multiple streams, delegate
its work to multiple controllers, but that fact is not visible
to LibvpxVp8Encoder.
This CL also squashes CL #126046 (Send notifications of RTT and
PLR changes to Vp8FrameBufferController) into it.
Bug: webrtc:10382
Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27206}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: Iedb3be944828a1caba55bbbd4dc0b56c55bbb7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127624
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27123}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
This CL plumbs an additional signal from VideoSendStream down to
VideoStreamEncoder, namely the amount of headroom that's left between
the encoder max bitrate and the current bitrate allocation for the
media track.
This will be used in follow-up CLs to tune encoder rate adjustment
and some codec specific paramaters a bit differently, based on the
knowledge if we are network constrained or not.
Bug: webrtc:10155
Change-Id: Ic6ccc79be5c6845468bab65b4ca9918b56923fa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125981
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27008}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/123920
Patch set 1 is identical to the previous CL, additional patch sets fix
the bug that was introduced and adds test coverage.
Since this "data base" only holds a single encoder instance it just
serves to confuse object ownership. Removing it and giving ownership
of generic encoder instance to VideoStreamEncoder.
This CL also removes VideoSender interface from video_coding_impl.h,
which is mostly a leftover from
https://webrtc-review.googlesource.com/c/src/+/123540
Bug: webrtc:10164
Change-Id: Ieaf23457d69af0d6356b70461112892b14760b19
Reviewed-on: https://webrtc-review.googlesource.com/c/124488
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26857}
This reverts commit 715c4765b1.
Reason for revert: Breaks WebRTC roll to Chromium.
https://chromium-review.googlesource.com/c/chromium/src/+/1484629
# Fatal error in: ../../third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc, line 796
# last system error: 0
# Check failed: diff_ms >= static_cast<int64_t>(0) (-307 vs. 0)
#
Original change's description:
> Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder
>
> Since this "data base" only holds a single encoder instance it just
> serves to confuse object ownership. Removing it and giving ownership
> of generic encoder instance to VideoStreamEncoder.
>
> This CL also removes VideoSender interface from video_coding_impl.h,
> which is mostly a leftover from
> https://webrtc-review.googlesource.com/c/src/+/123540
>
> Bug: webrtc:10164
> Change-Id: I9b7fec940dbcbccf3aa1278c2555da3bd5169ae1
> Reviewed-on: https://webrtc-review.googlesource.com/c/123920
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26835}
TBR=brandtr@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: I5432878c4c2e497cd848c4ce1b190e0307df03ca
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/124402
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26841}
Since this "data base" only holds a single encoder instance it just
serves to confuse object ownership. Removing it and giving ownership
of generic encoder instance to VideoStreamEncoder.
This CL also removes VideoSender interface from video_coding_impl.h,
which is mostly a leftover from
https://webrtc-review.googlesource.com/c/src/+/123540
Bug: webrtc:10164
Change-Id: I9b7fec940dbcbccf3aa1278c2555da3bd5169ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/123920
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26835}
This CL moves the functionality in VideoSender into VideoStreamEncoder
and simplifies the code where possible, given what we know of the
encoder state and that we now run on the encoder queue.
The intent here is to make it easier to remove the next parts, the
encoder database and generic encoder wrapper.
Bug: webrtc:10164
Change-Id: I8c108ccbe5db97cd9fd1e84228134709af845ea3
Reviewed-on: https://webrtc-review.googlesource.com/c/123540
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26813}
Reland with fixes for failing chromium tests.
Propagate VideoFrame::UpdateRect to encoder
Accumulate it in all places where frames can be dropped before they reach the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occasion then configuration have changed.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Bug: webrtc:10310
Change-Id: I18be73f47f227d6392bf9cb220b549ced225714f
Reviewed-on: https://webrtc-review.googlesource.com/c/123230
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26738}
Accumulate it in all places where frames can be dropped before they reach
the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occusion then
configuration have changed.
Bug: webrtc:10310
Change-Id: I2813ecd009eb730bd99ffa0a02f979091b56bf80
Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26711}
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}
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}
Currently, CPU overuse settings for HW encoders are sometimes being used
even though the actual encoder is a SW encoder, e.g. in case of SW fallback
when the encoder is initialized. Polling is_hardware_accelerated after the
encoder has been created and initialized will improve choosing the correct
CPU overuse settings.
Bug: webrtc:10065
Change-Id: Ic6bd67630a040b5a121c13fa63dd074006973929
Reviewed-on: https://webrtc-review.googlesource.com/c/116688
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26266}
This reverts commit 6613f8e98a.
Reason for revert: This change seemed innocent after all, so undoing speculative revert.
Original change's description:
> Revert "Refactor and remove media_optimization::MediaOptimization."
>
> This reverts commit 07276e4f89.
>
> Reason for revert: Speculative revert due to downstream crashes.
>
> Original change's description:
> > Refactor and remove media_optimization::MediaOptimization.
> >
> > This CL removes MediaOptmization and folds some of its functionality
> > into VideoStreamEncoder.
> >
> > The FPS tracking is now handled by a RateStatistics instance. Frame
> > dropping is still handled by FrameDropper. Both of these now live
> > directly in VideoStreamEncoder.
> > There is no intended change in behavior from this CL, but due to a new
> > way of measuring frame rate, some minor perf changes can be expected.
> >
> > A small change in behavior is that OnBitrateUpdated is now called
> > directly rather than on the next frame. Since both encoding frame and
> > setting rate allocations happen on the encoder worker thread, there's
> > really no reason to cache bitrates and wait until the next frame.
> > An edge case though is that if a new bitrate is set before the first
> > frame, we must remember that bitrate and then apply it after the video
> > bitrate allocator has been first created.
> >
> > In addition to existing unit tests, manual tests have been used to
> > confirm that frame dropping works as expected with misbehaving encoders.
> >
> > Bug: webrtc:10164
> > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26147}
>
> TBR=nisse@webrtc.org,sprang@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10164
> Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
> Reviewed-on: https://webrtc-review.googlesource.com/c/116780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26191}
TBR=nisse@webrtc.org,sprang@webrtc.org
Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/116960
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26213}
This reverts commit 07276e4f89.
Reason for revert: Speculative revert due to downstream crashes.
Original change's description:
> Refactor and remove media_optimization::MediaOptimization.
>
> This CL removes MediaOptmization and folds some of its functionality
> into VideoStreamEncoder.
>
> The FPS tracking is now handled by a RateStatistics instance. Frame
> dropping is still handled by FrameDropper. Both of these now live
> directly in VideoStreamEncoder.
> There is no intended change in behavior from this CL, but due to a new
> way of measuring frame rate, some minor perf changes can be expected.
>
> A small change in behavior is that OnBitrateUpdated is now called
> directly rather than on the next frame. Since both encoding frame and
> setting rate allocations happen on the encoder worker thread, there's
> really no reason to cache bitrates and wait until the next frame.
> An edge case though is that if a new bitrate is set before the first
> frame, we must remember that bitrate and then apply it after the video
> bitrate allocator has been first created.
>
> In addition to existing unit tests, manual tests have been used to
> confirm that frame dropping works as expected with misbehaving encoders.
>
> Bug: webrtc:10164
> Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26147}
TBR=nisse@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10164
Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
Reviewed-on: https://webrtc-review.googlesource.com/c/116780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26191}
This CL removes MediaOptmization and folds some of its functionality
into VideoStreamEncoder.
The FPS tracking is now handled by a RateStatistics instance. Frame
dropping is still handled by FrameDropper. Both of these now live
directly in VideoStreamEncoder.
There is no intended change in behavior from this CL, but due to a new
way of measuring frame rate, some minor perf changes can be expected.
A small change in behavior is that OnBitrateUpdated is now called
directly rather than on the next frame. Since both encoding frame and
setting rate allocations happen on the encoder worker thread, there's
really no reason to cache bitrates and wait until the next frame.
An edge case though is that if a new bitrate is set before the first
frame, we must remember that bitrate and then apply it after the video
bitrate allocator has been first created.
In addition to existing unit tests, manual tests have been used to
confirm that frame dropping works as expected with misbehaving encoders.
Bug: webrtc:10164
Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
Reviewed-on: https://webrtc-review.googlesource.com/c/115620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26147}
This reverts commit cdc5eb0de1.
Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().
Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
>
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
>
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}
TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.
Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
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}
Also renaming it Vp8TemporalLayers to show that it is codec specific.
Bug: webrtc:9012
Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f
Reviewed-on: https://webrtc-review.googlesource.com/c/104643
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25137}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
This is a reland of 529d0d9795
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}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
This reverts commit ced5cfdb35.
Reason for revert: Breaks downstream project.
Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
>
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
>
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}
TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.
Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".
It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.
Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
This is a followup to cl https://webrtc-review.googlesource.com/71380,
which reworked the way encoder resilience is done, and made the
nack_enabled flag unused.
Bug: webrtc:8830
Change-Id: I3de2508c97bc71e01c8f2232d16cd1f33e57fe4a
Reviewed-on: https://webrtc-review.googlesource.com/69986
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23080}
This deletes the resilienceOn flag in VideoCodecVP8 and VideoCodecVP9.
Instead, the implementations of VP8 and VP9 set resilience mode
internally, based on the configuration of temporal and spatial layers.
The nack_enabled argument to VideoCodecInitializer::SetupCodec becomes
unused with this cl. In a followup, it will be deleted, together with
the corresponding argument to VideoStreamEncoder methods.
An applications which really wants to configure resilience differently
can do that by injecting an EncoderFactory with encoders behaving
as desired.
Bug: webrtc:8830
Change-Id: I9990faf07d3e95c0fb4a56fcc9a56c2005b4a6fa
Reviewed-on: https://webrtc-review.googlesource.com/71380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23025}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
We only support on (formely kResilientStream) and off (formely
kResilienceOff). The third mode, kResilientFrames, was not
implemented.
Bug: None
Change-Id: Ida82f6a33eda9d943ea70bc8ae4e6bddb720b0e8
Reviewed-on: https://webrtc-review.googlesource.com/71481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22984}
This is a reland of fb82fcc7f9
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This reverts commit fb82fcc7f9.
Reason for revert: Appears to break Chromium, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Linux%20Tester/43756, where remoting_unittests failed.
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Change-Id: I47ee3ac42e62472d825a08c98e28f9ae53ec9fff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/70600
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22914}
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.
Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
This is a reland of dcc7e88cc7
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
Bug: webrtc:8415
Change-Id: I0ea7d4d679e7845907cfbe9a120f128ff2180e4b
Reviewed-on: https://webrtc-review.googlesource.com/68580
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22810}
