This CL adds miscellaneous unit tests for the
ResourceAdaptationProcessor, the VideoSourceRestrictions comparators and
the VideoStreamInputStateProvider.
Bug: webrtc:11172
Change-Id: If95f69644aaf2b43e3b19d5729bedef0b438c77b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174101
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@{#31147}
This makes it optional for an echo detector to report metrics through
the getStats interface.
Bug: webrtc:11539
Change-Id: I1fef93b7bf534637b69c16971d38709b3e849a08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31146}
Currently, ScopedJavaGlobalRef can only be set at creation and never
changed. This CL makes it possible to re-set these.
Bug: b/153389044
Change-Id: I6be92dae83a9f5f3d87aa44dde226b874f4ca0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31145}
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}
Implemented an analogue of the cpu_usage metrics from third_party/webrtc/video/video_analyzer.h for third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
Bug: webrtc:11496
Change-Id: Ifdc9daa3351f1df5db98beb8f7dc7156fc7c2a16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174020
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31141}
This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).
QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.
The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.
Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31140}
This introduces a function AudioProcessingImpl::SetCreateOptionalSubmodulesForTesting to simulate the exclusion of build-optional submodules, and tests of the currently only excludable submodule.
Bug: webrtc:11292
Change-Id: If492606205c9fdc669a6dce3a8989a434aeeed1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173746
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31138}
We'll need to deprecate the previous classes due to being used externally
as an API.
Bug: webrtc:11489
Change-Id: I64de29c8adae304d0b7628e24dd0abc5be6387ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31136}
This patch adds a field trial string for the IceController
to the factory interface, the string is from the
"WebRTC-IceControllerFieldTrials" key.
This makes it possible to add new field trials
using that key as needed.
Bug: chromium:1024965
Change-Id: I50498e45da3c49b8e1d620c90c674eedc15dc16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31134}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
WebRTC filenames use underscores to separate words so the ssl roots
file is rtc_base/ssl_roots.h instead of rtc_base/sslroots.h.
Bug: chromium:978779
Change-Id: I2fa11c38a566e177775deb3d42230d956efc8ccc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31132}
This is important when we have multiple named devices connected over
USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
choose a specific input device to route from.
Bug: b/154440591
Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
Commit-Queue: Robin Lee <rgl@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31130}
Move creation of video sinks for dumping and showing rendered video on
screen into video quality analyzer injection helper to eliminate need
to search for video config in on track callback, which makes this more
reliable and reusable.
Bug: webrtc:11479
Change-Id: I6bb5409688fd39268f9f97bde4c9b0833a64396b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31128}
Reset the frame transformer delegate's callback to
ChannelReceive::OnReceivedPayloadData when the channel is destroyed, to
prevent future callbacks from the delegate.
Bug: chromium:870644
Change-Id: Iaa2c1b7b26dc38709d3fe64a180ccc6a60a1ec9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173751
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31121}
We have observed an internal deadlock in libGLESv2_adreno where one
thread is in eglCreateContext and another thread in glUseProgram. We
have observed similar deadlocks before and started to synchronize all
access to the offending GL functions. Calls to eglCreateContext are
already synchronized, and this CL synchronizes calls to glUseProgram as
well.
Bug: b/153513005
Change-Id: I576e564aab44c9e429f2b1407105ed72942c309e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173742
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31118}
This is based on channel_receive.cc implementation where non-audio
related logics are trimmed off for smaller footprint in size.
Bug: webrtc:11251
Change-Id: I743c9f93f509fa6fcc12981fa73a6f01ce38348c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172821
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31117}
This CL is functionally a noop but may reduce thread wakupes in some
cases.
In particular, consider a send task scheduled for time T. While waiting
for that, a higher-priority packet than the top of the current queue is
added (e.g. an audio packet), and a send is executed immediately.
After sending, it resets the field indicating that a scheduled task is
expected at time T. It then polls NextSendTime() and schedules a new
task, likely at or very close to T. Causing unnecessary task queue
churn and behavior that is more difficult to reason about.
Bug: webrtc:10809
Change-Id: Ic5706f2cc06df3f27cc3e7b473d4de29a669473b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173700
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31116}
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}
This caused at least one trial in RTPSender not to be properly parsed.
This CL also updates RtpVideoSender and RtpPayloadParams to use
WebRtcKeyValueConfig instead of the static field_trial methods, in
order to facilitate injectable behavior in the future.
Bug: webrtc:11508
Change-Id: I995939bd3e7c2f81e5050383c3e4daf933498520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173705
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31108}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
This gets to the heart of unblocking call-level adaptation, largely
made possible due to the previous CLs in the chain.
The parts of the code that are responsible for responding to resource
usage signals, obtaining adaptations and applying them are moved to
ResourceAdaptationProcessor in call/adaptation/.
The parts of the code that are responsible for managing
VideoStreamEncoder-specific resources stay inside the
VideoStreamEncoderResourceManager class in video/adaptation/.
After this CL lands it should soon be possible to move the Processor
over to a separate task queue and let the Manager stay on the encoder
queue if PostTasks are added for communication between the two objects.
Bug: webrtc:11172
Change-Id: Ifa212467b4afd16e7ebfb9adfe17d2dca1cb7d67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173021
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@{#31105}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
By pushing VideoAdaptationCounters updates on VideoSourceRestrictions
changes, alongside the Resource* that triggered the adaptation, we are
able to update |active_counts_| without an explicit dependency on the
VideoStreamAdapter. This allows a future CL to split up "processor"
logic from "video stream encoder resource and active counts" logic,
which will ultimately be necessary in order to do processing on a
"processing queue" and encoder and stats logic on the "encoder queue".
If the restrictions got cleared by an API call
(ResetVideoSourceRestrictions() or SetDegradationPreference()) we pass
null as the "reason_resource". This allows is to clear the
active_counts_, and the code that invokes
OnVideoSourceRestrictionsUpdated() does not have to be aware of
active_counts_ (needed to split the processor module in two).
Bug: webrtc:11172
Change-Id: Icab6d5121c0ebd27d2a00f1bffc8191f8f05f562
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173000
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@{#31103}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
The VideoStreamAdapter is currently responsible for aborting and not
providing adaptations if we are bitrate constrained
(kIsBitrateConstrained). Whether or not we are bitrate constrained is
clearly a resource question and should be phrased as such. By moving
this logic to Resource::IsAdaptationUpAllowed(), the VideoStreamAdapter
can continue to be thread-agnostic when a future CL introduces a
"processing queue", and the VideoStreamAdapter can be simplified: it
returns Adaptations even if we are constrained (but we refuse to Apply
them any resource rejects it).
This CL adds new Resource classes as inner classes of
ResourceAdaptationProcessor that take on the responsibility of
kIsBitrateConstrained logic:
PreventIncreaseResolutionDueToBitrateResource and
PreventAdaptUpInBalancedResource.
A third class, PreventAdaptUpDueToActiveCounts, also allows us to move
adaptation-aborting logic. This piece of code appears to be about not
adapting up if we’re already at the highest setting, which would be
VideoStreamAdapter responsibility (covered by
Adaptation::Status::kLimitReached), but it is actually more complicated
than that: the active_counts_ care about "reason", so it is really about
"is this resource type OK with you adapting up?". We should probably
rewrite this code in the future, but for now it is moved to an inner
class of ResourceAdaptationProcessor.
Other misc changes:
- ApplyDegradationPreference is moved to video_stream_adapter.[h/cc]
and renamed "Filter".
- OnResourceOveruse/Underuse now use Resource* as the reason instead of
AdaptReason. In a future CL, the processor will be split into a
"processor" part and a "video stream encoder resource manager" part.
Only the manager needs to know about AdaptReason since this is only
used for |active_counts_| and we want to get rid of it as much as
possible as it is not future-proof.
Bug: webrtc:11172
Change-Id: I2eba9ec3d717f7024c451aeb14635fe759551318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172930
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@{#31099}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
The "input state" of a VideoStream, needed for adaptation and
decision-making, are: source resolution and frame rate, codec type and
min pixels per frame (based on encoder scaling settings). These values
are modified on the encoder queue of the VideoStreamEncoder.
But in order to unblock call-level adaptation processing, where
adaptation and decision making happens off the encoder queue, a snapshot
of the input states need to be available at point of processing:
introducing the VideoStreamInputState.
In this CL, the VideoStreamInputStateProvider is added to provide input
state snapshots across threads based on input from VideoStreamEncoder
and VideoStreamEncoderObserver.
The input state's HasInputFrameSizeAndFramesPerSecond() can now be
DCHECKed inside the VideoStreamAdapter in favor of having less
Adaptation::Status codes. Whether input is "sufficient" for adaptation
is now the responsibility of the Processor. (Goal: adapter is purely a
Adaptation generator and apply-er.)
Somewhat tangental, this CL also deletes VideoStreamEncoder-specific
methods from ResourceAdaptationProcessorInterface making them an
implementation detail of ResourceAdaptationProcessor. In a future CL,
the "processor" will be split up into a "processor" part and a "video
stream encoder resource manager" part - more on that later.
Bug: webrtc:11172
Change-Id: Id9b158f569db0140b75360aaf0f7e2e28fb924f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172928
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@{#31098}
