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}
https://webrtc-review.googlesource.com/c/src/+/172847
------------ original description --------------
Preparation for ReceiveStatisticsProxy lock reduction.
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: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
Save the frame transformer set on unsignaled receivers, and set the
transformer when the ssrc becomes known.
Pass the receiver's ssrc on registering the transformed frame callback,
to associate separate frame transformer sinks for each receiver.
Bug: chromium:1065838
Bug: chromium:1065838
Change-Id: I2a214bdb6cb9a8012928a03f046f311c344370f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173201
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31051}
The RtpVideoStreamReceiverFrameTransformerDelegate::IsKeyFrame()
implementation was relying on the EncodedFrame::is_keyframe() API, which
checks the number of references to a frame. However the number of
references is not updated until after the frame is managed by the
receiver, after the frame has been transformed.
Update the delegate's IsKeyFrame() implementation to use the type of the
frame instead. The frame type is updated before transforming the frame,
on parsing the generic descriptor.
Bug: chromium:1068468
Change-Id: I84dadaecb1cd485262c2f1681dfa653d84693f69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173025
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31024}
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.
Reason for revert: Speculative revert: breaks downstream project
Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
>
> 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}
TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
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}
