Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
Add a new frame interface to be used by frame transformers in Insertable
Streams. TransformableFrameInterface will replace
video_coding::EncodedFrame in a follow up CL, once downstream
dependecies are updated to use the new interface.
Until the functions using video_coding::EncodedFrame are removed from
the API, the video sender and receiver frame transformer delegates call
both function versions to avoid breaking tests downstream.
The TransformableFrameInterface will be used for both audio and video
frame transformers in follow-up CLs.
Bug: webrtc:11380
Change-Id: I9389a8549c156e13b1d8c938ff51eaa69c502f33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171863
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30941}
This CL fixes a few issues where the reported fraction of frames
allocated to various temporal layers could be incorrect:
* In LibvpxVp8Encoder, calling GetEncoderInfo() while not initialized,
or when first configuring with temporal layers and then without,
could trigger incorrect fps allocations.
* In VP9 when different spatial layers have different max framerates,
the layer fps should be compared to the layer with the highest
configured fps, not codec_.maxFramerate which is updated to the
current input fps on SetRates().
* In EncoderBitrateAdjuster, just warn and ignore if a layer has
non-zero bps but zero fps, rather than passing down the chain and
risk weird behavior or divide by zero.
Bug: b/152040235
Change-Id: I548fb3e099b1ec9f536a7b93313fb40c4d32e596
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171516
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30880}
--- Background ---
The webrtc::VideoSendStream::StreamStats are converted into
VideoSenderInfo objects which turn into "outbound-rtp" stats objects in
getStats() (or "ssrc" objects in legacy getStats()).
StreamStats are created for each type of substream: RTP media streams,
RTX streams and FlexFEC streams - each with individual packet counters.
The RTX stream is responsible for retransmissions of a referenced media
stream and the FlexFEC stream is responsible for FEC of a referenced
media stream. RTX/FEC streams do not show up as separate objects in
getStats(). Only the media streams become "outbound-rtp" objects, but
their packet and byte counters have to include the RTX and FEC counters.
--- Overview of this CL ---
This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes
StreamStats of all kinds as input, and outputs media-only StreamStats
- incorporating the RTX and FEC counters into the relevant media
StreamStats.
The merged StreamStats objects is a smaller set of objects than the
non-merged counterparts, but when aggregating all packet counters
together we end up with exact same packet and count as before.
Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates
the StreamStats into a single VideoSenderInfo (single "outbound-rtp"),
this CL should not have any observable side-effects. Prior to this CL:
aggregate StreamStats. After this CL: merge StreamStats and then
aggregate them.
However, when simulcast stats are implemented (WIP CL:
https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media
stream should turn into an individual "outbound-rtp" object. We will
then no longer aggregate all StreamStats into a single "info". This CL
unblocks simulcast stats by providing StreamStats objects that could be
turned into individual VideoSenderInfos.
--- The Changes ---
1. Methods added to RtpConfig to be able to easily tell the relationship
between RTP, RTX and FEC ssrcs.
2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that
replaces the booleans (is_rtx, is_flexfec).
3. "referenced_media_ssrc" is added to StreamStats, making it possible
to tell which kRtx/kFlexFec stream stats need to be merged with which
kMedia StreamStats.
4. MergeInfoAboutOutboundRtpSubstreams() added and used.
Bug: webrtc:11439
Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30869}
In preparation for a change that rely on payload type beeing present.
As side effect, fix test related to RED payload type.
Bug: None
Change-Id: I42f8460fbec578184da058c1f6b9620d497d576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171508
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30864}
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay
(previously it was "only" network_id)
The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.
OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/
BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
this is a step towards resolving own todo: making AssembleFrame part of
the VideoRtpDepacketizer interface and replacing codec check with a
call to a virtual function.
RtpVideoStreamReceiver has access to the VideoRtpDepacketizers,
PacketBuffer - hasn't.
Bug: None
Change-Id: I83df09975c092bdb71bab270ced356d79a50683d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168056
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30833}
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}
This code was previously only exercised by
video_stream_adapter_unittest.cc and other tests, acting more like
integration tests than unit tests. Now that the VideoStreamAdapter is
in a good state, more extensive test coverage is added.
Testing includes:
- Default restrictions.
- Adapting up or down in "maintain-framerate", "maintain-resolution"
and "balanced", including...
- expecting how frame rate and/or resolution is affected,
- reaching kLimitReached,
- and reaching unrestricted.
- That "disabled" does not adapt.
- When adaptation is not possible, including...
- kInsufficientInput
- kAwaitingPreviousAdaptation
- kIsBitrateConstrained
- PeekNextRestrictions()
- "balanced" + "screenshare" = "maintain-resolution"
- Change degradation preference to/from "balanced" clears restrictions.
- That using invalidated adaptations triggers DCHECKs.
Bug: webrtc:11393
Change-Id: I28e2cf227bc1fd8871ee0d18d9570d4063449160
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170625
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@{#30816}
It was possible before if an input fps dropped due to cpu adaptation
Also, this CL removes occasional test failure (it could've happened if
input framerate got very low)
Bug: webrtc:11432,webrtc:11426
Change-Id: Id1a4df23302f7b8ab6781f1e7cca5112bfcfe9ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170469
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30802}
This CL introduces the Adaptation class used by VideoStreamRestrictor.
This refactors the AdaptationTarget, AdaptationTargetOrReason,
CannotAdaptReason and AdaptationAction.
What is publicly exposed is simply a Status code. If it's kValid then
we can adapt, otherwise the status code describes why we can't adapt
(just like CannotAdaptReason prior to this CL). This means
AdaptationTargetOrReason is no longer needed. Target+reason are merged.
The other classes are renamed and moved and put in the private
namespace of Adaptation: Only the VideoStreamAdapter (now a friend
class of Adaptation) and its inner class VideoSourceRestrictor needs to
know how to execute the adaptation.
Publicly, you can now tell the effects of the adaptation without
applying it with PeekNextRestrictions() - both current and next steps
are described in terms of VideoSourceRestrictions. The rest are hidden.
This would make it possible, in the future, for a Resource to accept or
reject a proposed Adaptation by examining the resulting frame rate and
resolution described by the resulting restrictions. E.g. even if we are
not overusing bandwidth at the moment, the BW resource can prevent us
from applying a restriction that would exceed the BW limit before we
apply it.
This CL also moves input to a SetInput() method, and Increase/Decrease
methods of VideoSourceRestrictor are made private in favor of
ApplyAdaptationSteps().
Bug: webrtc:11393
Change-Id: Ie5e2181836ab3713b8021c1a152694ca745aeb0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170111
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@{#30794}
These are considered expandable, and since video tests are very
expensive (45s each), let's remove them.
Bug: webrtc:11426
Change-Id: I4aea18e93d3b3672900650aacf0b5524c52c2900
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170364
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30790}
This introduces AdaptationTargetOrReason and gets rid of
VideoStreamAdapter's dependency on the VideoStreamEncoderObserver.
AdaptationTargetOrReason provides information about why an adaptation
target could not be returned from GetAdaptUpTarget() and
GetAdaptDownTarget() with the enum CannotAdaptReason and the boolean
min_pixel_limit_reached.
While the enum value is not used by the caller in this CL, it makes
explicit reasons the adapter is allowed to reject a target. TODOs are
added documenting how we want to get rid of kAwaitingPreviousAdaptation
for multi-stream use cases and how kIsBitrateConstrained can be
rephrased as a resource problem in the future.
min_pixel_limit_reached() allows us to move the responsibility of stats
reporting to the module. A TODO documents how this could be replaced by
kLimitReached or similar logic in the future.
Bug: webrtc:11393
Change-Id: Iffdd8ddb01641937741fac353174ea14168477ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169928
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@{#30789}
The dependencies have been updated to use the new constructor.
Bug: webrtc:11380
Change-Id: I1ded1816b94fd069d729df50ff83842eca054acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170223
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30766}
The EffectiveDegradationPreference() exposed an "implementation detail"
of the VideoStreamAdapter - how degradation preference may be modified.
By changing the return value of ApplyAdaptationTarget() this dependency
could be removed. We still have a TODO to get rid of the
ResourceListenerResponse enum, but that is QualityScaler related work.
This CL does the following:
- Module's GetAdaptUpTarget/GetAdaptDownTarget/ApplyAdaptationTarget
methods are removed in favor if invoking the VideoStreamAdapter's
version of these methods directly.
- Removing the EffectiveDegradationPreference() usage in
OveruseFrameDetectorResourceAdaptationModule meant moving that usage
to VideoStreamAdapter.
- MinPixelsPerFrame() is moved to VideoStreamAdapter; this is "can
adapt?" logic, i.e. the adapter's responsibility.
Bug: webrtc:11393
Change-Id: I75091ce97093bfa48a6d883492de30ed4b004492
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169859
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@{#30714}
This CL moves GetAdaptUpTarget(), GetAdaptDownTarget() and
ApplyAdaptationTarget() - and related code - to the VideoStreamAdapter.
This includes pieces related to calculating how to adapt, including:
- DegradationPreference
- BalancedDegradationPreference
- AdaptationRequest and last_adaptation_request_
- CanAdaptUpResolution()
The VideoStreamAdapter's interface has changed: VideoSourceRestrictor
methods are now hidden in favor of methods exposing AdaptationTarget.
This CL also does some misc moves:
- GetEncoderBitrateLimits is moved and renamed to
VideoEncoder::EncoderInfo::GetEncoderBitrateLimitsForResolution.
- EncoderSettings moved to a separate file.
// For api/video_codecs/video_encoder.[cc/h] changes, which is the
// moving of a function.
TBR=sprang@webrtc.org
Bug: webrtc:11393
Change-Id: Ie6bd8ef644ce927d7eca6ab90a0a7bcace682f3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169842
Reviewed-by: Henrik Boström <hbos@webrtc.org>
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@{#30708}
This CL simply moves the VideoSourceRestrictor from being an inner class
of OveruseFrameDetectorResourceAdaptationModule to a new class,
VideoStreamAdapter.
In follow-up CLs, the responsibility of determining what the next step
for adapting up or down should also be moved to the VideoStreamAdapter.
The end-goal is that the VideoStreamAdapter takes care of "can adapt?"
and "do adapt!" type of logic so that a multi-stream aware adaptation
module can decide which stream (adapter) to adapt, and the adapter can
take care of the nitty gritty details of doing so.
In this CL the "can?"/"do!" part is realized but not the logic for
determining what the next step up or down is, and the class interface
needs improvement.
This CL also sets up the video/adaptation/ subdirectory and moves the
AdaptationCounters class here. Other adaptation-related classes (e.g.
the module and its resources) should move into this directory as well
in the future.
Bug: webrtc:11393
Change-Id: I2c12c1281eca854c62791abb65f0aca47a119726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169542
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30705}
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}
When selecting all streams there was an index out of bounds
checking the selected temporal layer, which is -1 so was irrelevant.
My fix is to prevent selecting a temporal layer and all streams
at the same time.
Bug: webrtc:11402
Change-Id: I0641b926cba35878945b866f2c59b4b0281f0852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169720
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30679}
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.
The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
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}
Add a new API in RTPSenderInterface, to be called from the browser side
to insert a frame transformer between the Encoded and the Packetizer.
The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in RTPSenderVideo, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169128.
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30642}
The fact that they weren't const is probably a remenant of the good ol'
days this class being multi-threaded and having to acquire mutexes. Now
they can properly be made const.
In order to make GetConstAdaptCounter() const, this CL makes sure a
cleared adapt_counters_ map contains mappings for all degradation
preferences to default-constructed AdaptCounters. Previously, if the
mapping was missing it was implicitly inserted by
GetConstAdaptCounter(). Now it can DCHECK that mappings always exists
instead, and it always has something to return.
Bug: webrtc:11222
Change-Id: If33227fe6572eb1d6cc6b9f851d6d174d035c110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168953
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@{#30611}
* 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}
Today OnResourceOveruse() and OnResourceUnderuse() implicitly checks
preconditions and if they pass calculate the next target, and if those
are usable it applies them to the VideoSourceRestrictions. These are two
big "MaybeAdapt" methods.
This CL takes us one step closer to "GetNextTarget", "CanApplyTarget?"
and "DoApplyTarget!"-logic, which will allow us to more easily evaluate
a multitude of alternative configurations and decide which one to pick
(e.g. multi-stream adaptation).
But it does not take us all the way there. In this CL we have:
- CanAdaptUp, CanAdaptDown: This covers *most* of the preconditions.
- OnResourceUnderuse, OnResourceOveruse: This aborts if CanAdapt returns
false. If they pass, we calculate the next target and maybe-adapt it.
This is roughly outlined in document still in draft:
https://docs.google.com/document/d/1YMg-AycFZR1DS6hEav9OzJ3hqxFil09qPhlTAgQrU1g/edit?usp=sharing.
A future CL should make the target more explicit and we should know if
the target can be applied before we even try.
Bug: webrtc:11222
Change-Id: If18d9572884aa6ba2350e4670a1516da5835cc98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168721
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@{#30605}
Adapting up or down is currently a "Maybe Adapt" method. In the future
we will want to be able to decide which stream to adapt, and as such we
need to be able to tell if a stream is able to do so.
This takes us one step in that direction, by refactoring
OveruseFrameDetectorResourceAdaptationModule::VideoSourceRestrictor's
methods to follow a simple pattern:
- What is the next step?
GetHigherFrameRateThan, GetLowerFrameRateThan,
GetHigherResolutionThan, GetLowerResolutionThan
- Can we adapt?
CanIncreaseFrameRate, CanDecreaseFrameRate,
CanIncreaseResolution, CanDecreaseResolution
- Do adapt!
IncreaseFrameRateTo, DecreaseFrameRateTo,
IncreaseResolutionTo, DecreaseResolutionTo
Hopefully this makes the code easier to follow.
This CL changes the "Request Higher/Lower" methods to take the target
as input instead of implicitly calculating the target from the current
input resolution or frame rate.
Bug: webrtc:11222
Change-Id: If625834e921a24a872145105f5d553fb8f9f1795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168966
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@{#30600}
