This reverts commit 32ca95145c.
Reason for revert: Internal test failure
Original change's description:
> Only enable conference mode simulcast allocations with flag enabled
>
> Non-conference mode simulcast screenshares were mistakenly using the
> conference mode semantics in the simulcast rate allocator, which broke
> spec compliant usage in some situation.
>
> This behavior should only be used when explicitly using the SDP entry
> "a=x-google-flag:conference" in both offer and answer.
>
> Bug: webrtc:11310, chromium:1093819
> Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31828}
TBR=ilnik@webrtc.org,hta@webrtc.org,orphis@webrtc.org
Change-Id: I5ccb6e87594f491ba09fe6b837ee24d63db878ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11310
Bug: chromium:1093819
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180801
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31829}
Non-conference mode simulcast screenshares were mistakenly using the
conference mode semantics in the simulcast rate allocator, which broke
spec compliant usage in some situation.
This behavior should only be used when explicitly using the SDP entry
"a=x-google-flag:conference" in both offer and answer.
Bug: webrtc:11310, chromium:1093819
Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31828}
This is a reland of 003c9be817
Found some downstream code that relies on
NetworkMonitorFactory::SetFactory, so I'm adding those methods back
temporarily. BasicNetworkManager will fall back to the static factory
if the one passed into PeerConnectionFactory is null.
Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}
TBR=hta@webrtc.org, sakal@webrtc.org
Bug: webrtc:9883
Change-Id: I2e817c423f21936f87532a9694eb9a0a1b70c212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180722
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31824}
This reverts commit 7ded733518.
Reason for revert: Found more code calling NetworkMonitorFactory::SetFactory...
Original change's description:
> Reland "Pass NetworkMonitorFactory through PeerConnectionFactory."
>
> This is a reland of 003c9be817
>
> Original change's description:
> > Pass NetworkMonitorFactory through PeerConnectionFactory.
> >
> > Previously the instance was set through a static method, which was
> > really only done because it was difficult to add new
> > PeerConnectionFactory construction arguments at the time.
> >
> > Now that we have PeerConnectionFactoryDependencies it's easy to clean
> > this up.
> >
> > I'm doing this because I plan to add a NetworkMonitor implementation
> > for iOS, and don't want to inherit this ugliness.
> >
> > Bug: webrtc:9883
> > Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31815}
>
> TBR=hta@webrtc.org, sakal@webrtc.org
>
> Bug: webrtc:9883
> Change-Id: Ibf69a22e8f94226908636c7d50ff9eda65bd4129
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180720
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31822}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,hta@webrtc.org
Change-Id: Iae51b94072cec9abc021eed4e51d1fbeee998adc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180721
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31823}
This is a reland of 003c9be817
Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}
TBR=hta@webrtc.org, sakal@webrtc.org
Bug: webrtc:9883
Change-Id: Ibf69a22e8f94226908636c7d50ff9eda65bd4129
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180720
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31822}
Network emulation layer provides per source split for incoming stats for
endpoint. Do the same for outgoing stats per destination.
Bug: webrtc:11756
Change-Id: I2369ae8906546c27133273b1be17ce74c253c6e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180500
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31820}
This reverts commit 003c9be817.
Reason for revert: Breaks downstream build which is still using
SetFactory/ReleaseFactory. Probably will need to update this in lockstep.
Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,hta@webrtc.org
Change-Id: I1f09df7be9c860017d515e5a87488340afa6eda6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180640
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31818}
as it has been used in production for a while already.
BUG=None
Change-Id: Idd466f3ad02129a8125e54122e77036959e67aa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175131
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31817}
Previously the instance was set through a static method, which was
really only done because it was difficult to add new
PeerConnectionFactory construction arguments at the time.
Now that we have PeerConnectionFactoryDependencies it's easy to clean
this up.
I'm doing this because I plan to add a NetworkMonitor implementation
for iOS, and don't want to inherit this ugliness.
Bug: webrtc:9883
Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31815}
This is a reland of d4089cae47
with the following fix:
Invoke MaybeStartGathering as the last step of DoSetLocalDescription.
This ensures that candidates and onicegatheringstatechange does not
happen before SLD is resolved. This is important for passing
external/wpt/webrtc/RTCPeerConnection-iceGatheringState.html.
Original change's description:
> [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
>
> BACKGROUND
>
> When SLD is invoked with SetSessionDescriptionObserver, the observer is
> called by posting a message back to the execution thread, delaying the
> call. This delay is "artificial" - it's not necessary; the operation is
> already complete. It's a post from the signaling thread to the signaling
> thread. The rationale for the post was to avoid the observer making
> recursive calls back into the PeerConnection. The problem with this is
> that by the time the observer is called, the PeerConnection could
> already have executed other operations and modified its states.
>
> This causes the referenced bug: one can have a race where SLD is
> resolved "too late" (after a pending SRD is executed) and the signaling
> state observed when SLD resolves doesn't make sense.
>
> When implementing Unified Plan, we fixed similar issues for SRD by
> adding a version that takes SetRemoteDescriptionObserverInterface as
> argument instead of SetSessionDescriptionObserver. The new version did
> not have the delay. The old version had to be kept around not to break
> downstream projects that had dependencies both on he delay and on
> allowing the PC to be destroyed midst-operation without informing its
> observers.
>
> THIS CL
>
> This does the old SRD fix for SLD as well: A new observer interface is
> added, SetLocalDescriptionObserverInterface, and
> PeerConnection::SetLocalDescription() is overloaded. If you call it with
> the old observer, you get the delay, but if you call it with the new
> observer, you don't get a delay.
>
> - SetLocalDescriptionObserverInterface is added.
> - SetLocalDescription is overloaded.
> - The adapter for SetSessionDescriptionObserver that causes the delay
> previously only used for SRD is updated to handle both SLD and SRD.
> - FakeSetLocalDescriptionObserver is added and
> MockSetRemoteDescriptionObserver is renamed "Fake...".
>
> Bug: chromium:1071733
> Change-Id: I920368e648bede481058ac22f5b8794752a220b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31798}
TBR=hta@webrtc.org
Bug: chromium:1071733
Change-Id: Ic6e8d96afa1c19604762f373716c08dbfa9d178c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180481
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31804}
This reverts commit d4089cae47.
Reason for revert: Breaks chromium WPT that is timing sensitive to onicegatheringstatechanges.
This CL accidentally moved the MaybeStartGatheringIceCandidates to after completing the SLD call. The fix is to move it back. I'll do that in a re-land.
Original change's description:
> [Perfect Negotiation] Implement non-racy version of SetLocalDescription.
>
> BACKGROUND
>
> When SLD is invoked with SetSessionDescriptionObserver, the observer is
> called by posting a message back to the execution thread, delaying the
> call. This delay is "artificial" - it's not necessary; the operation is
> already complete. It's a post from the signaling thread to the signaling
> thread. The rationale for the post was to avoid the observer making
> recursive calls back into the PeerConnection. The problem with this is
> that by the time the observer is called, the PeerConnection could
> already have executed other operations and modified its states.
>
> This causes the referenced bug: one can have a race where SLD is
> resolved "too late" (after a pending SRD is executed) and the signaling
> state observed when SLD resolves doesn't make sense.
>
> When implementing Unified Plan, we fixed similar issues for SRD by
> adding a version that takes SetRemoteDescriptionObserverInterface as
> argument instead of SetSessionDescriptionObserver. The new version did
> not have the delay. The old version had to be kept around not to break
> downstream projects that had dependencies both on he delay and on
> allowing the PC to be destroyed midst-operation without informing its
> observers.
>
> THIS CL
>
> This does the old SRD fix for SLD as well: A new observer interface is
> added, SetLocalDescriptionObserverInterface, and
> PeerConnection::SetLocalDescription() is overloaded. If you call it with
> the old observer, you get the delay, but if you call it with the new
> observer, you don't get a delay.
>
> - SetLocalDescriptionObserverInterface is added.
> - SetLocalDescription is overloaded.
> - The adapter for SetSessionDescriptionObserver that causes the delay
> previously only used for SRD is updated to handle both SLD and SRD.
> - FakeSetLocalDescriptionObserver is added and
> MockSetRemoteDescriptionObserver is renamed "Fake...".
>
> Bug: chromium:1071733
> Change-Id: I920368e648bede481058ac22f5b8794752a220b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31798}
TBR=hbos@webrtc.org,hta@webrtc.org
Change-Id: Ie1e1ecc49f3b1d7a7e230db6d36decbc4cbe8c86
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1071733
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180480
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31802}
BACKGROUND
When SLD is invoked with SetSessionDescriptionObserver, the observer is
called by posting a message back to the execution thread, delaying the
call. This delay is "artificial" - it's not necessary; the operation is
already complete. It's a post from the signaling thread to the signaling
thread. The rationale for the post was to avoid the observer making
recursive calls back into the PeerConnection. The problem with this is
that by the time the observer is called, the PeerConnection could
already have executed other operations and modified its states.
This causes the referenced bug: one can have a race where SLD is
resolved "too late" (after a pending SRD is executed) and the signaling
state observed when SLD resolves doesn't make sense.
When implementing Unified Plan, we fixed similar issues for SRD by
adding a version that takes SetRemoteDescriptionObserverInterface as
argument instead of SetSessionDescriptionObserver. The new version did
not have the delay. The old version had to be kept around not to break
downstream projects that had dependencies both on he delay and on
allowing the PC to be destroyed midst-operation without informing its
observers.
THIS CL
This does the old SRD fix for SLD as well: A new observer interface is
added, SetLocalDescriptionObserverInterface, and
PeerConnection::SetLocalDescription() is overloaded. If you call it with
the old observer, you get the delay, but if you call it with the new
observer, you don't get a delay.
- SetLocalDescriptionObserverInterface is added.
- SetLocalDescription is overloaded.
- The adapter for SetSessionDescriptionObserver that causes the delay
previously only used for SRD is updated to handle both SLD and SRD.
- FakeSetLocalDescriptionObserver is added and
MockSetRemoteDescriptionObserver is renamed "Fake...".
Bug: chromium:1071733
Change-Id: I920368e648bede481058ac22f5b8794752a220b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31798}
It is a follow up CL to
https://webrtc-review.googlesource.com/c/src/+/179368.
Now when network stats became more complex structure it's better to hide
its implementation details and provide an interface for read-only
access.
Bug: webrtc:11756
Change-Id: I1980ef938f8de0c6aa90092d1dc90a14a82e0ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179840
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31784}
local_addresses is a list of IPs that were used to send data, which was
used during stats calculation.
Bug: webrtc:11756
Change-Id: Ie6307eaa69c73ebe9f69e44503752151be9e9ef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179841
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31774}
i.e. when chain are used,
require each decode target to be protected by some chain.
where previously it was allowed to mark decode target as unprotected.
See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125
Bug: webrtc:10342
Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31772}
So that applications can construct a default struct without naming the
members.
Bug: None
Change-Id: Idd9028bee9016670e776f17a62077eb9c34d6f2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179485
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31742}
This is a reland of 94fe0d3de5 with a fix.
Original change's description:
> Complete migration from "track" to "inbound-rtp" stats
>
> Bug: webrtc:11683
> Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Cr-Commit-Position: refs/heads/master@{#31683}
Bug: webrtc:11683
Change-Id: I173b91625174051c02ff34127aaf6c086d3c5c66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179060
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31696}
put default values next to the (named) members.
Remove constructor
declared as VideoQualityTestFixtureInterface::Param::Param,
but defined as VideoQualityTest::Param::Param
avoid braced initialisation for structures with lot's of fields that change over time
Bug: None
Change-Id: I647e3ace47ae981c515d6939274cb2744dd1b866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178811
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31674}
Add TimeController to the CreatePeerConnectionE2EQualityTestFixture
method as a first step to make PC level framework compatible with
TimeController abstraction.
Bug: webrtc:11743
Change-Id: I69305abc880059bf9fe1d4f2e3b7c10cf35417db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178485
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31607}
When enabled:
- Creates an audio network adapter config that is passed to audio send
stream.
- Configures a lower default min bitrate.
All parameters can be configured via a field trial that can also force
enable the audio network adaptor (this is mainly intended for testing).
Bug: chromium:1086942
Change-Id: I48dfcca1ee2948084199352abed6212a6c78eb6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177840
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31565}
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.
Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
This is needed because chromium build targets need to be exported for
its component builds.
// TBR because this is a purely building related change and it has been
// reviewed by mbonadei@.
TBR=stefan@webrtc.org
Bug: webrtc:11525
Change-Id: I97f0c814b11e7fad86eeff319e644ae51204c3b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31534}
This reverts commit 71db9acc40.
Reason for revert: breaks downstream project.
Reason for force push: win bot broken.
Original change's description:
> RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
>
> This change adds exposure of a new transceiver method for
> modifying the extensions offered in the next SDP negotiation,
> following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.
>
> Features:
> - The interface allows to control the negotiated direction as
> per https://tools.ietf.org/html/rfc5285#page-7.
> - The interface allows to remove an extension from SDP
> negotiation by modifying the direction to
> RtpTransceiverDirection::kStopped.
>
> Note: support for signalling directionality of header extensions
> in the SDP isn't implemented yet.
>
> https://chromestatus.com/feature/5680189201711104.
> Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
>
> Bug: chromium:1051821
> Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31487}
TBR=hta@webrtc.org,handellm@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: true
Bug: chromium:1051821
Change-Id: I70e1a07225d7eeec7480fa5577d8ff647eba6902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31516}
This unblocks injecting platform-specific resources, such as power
usage signals in Chrome.
This CL adds AddAdaptationResource to PeerConnectionInterface and
integration tests verifying that if an injected resource is overusing,
resolution will soon be reduced.
To aid testing, some testing-only classes have been updated.
Bug: webrtc:11525
Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31505}
This CL adds AddAdaptationResource to Call and
AddAdaptationResource/GetAdaptationResources method to relevant
VideoSendStream and VideoStreamEncoder interfaces and implementations.
Unittests are added to ensure that resources can be added to the Call
both before and after the creation of a VideoSendStream and that the
resources always gets added to the streams.
In a follow-up CL, we will continue to plumb the resources all the way
to PeerConnectionInterface, and an integration test will then be added
to ensure that injected resources are capable of triggering adaptation.
Bug: webrtc:11525
Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31499}
This allows const getters that query const state to be called without
marshalling calls between threads. This must not be used to
return pointers/references etc.
I'm starting by using this macro with the data channel which has a
few of these getters, as well as changing things a bit to make more
parts of the implementation, const.
Change-Id: I6ec7a3774cd8f7be2ef122fb7c7fc5919afee600
Bug: webrtc:11547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176846
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31489}
for TaskQueueTest.
This suit is instantiated in the different binary targets by design.
Bug: None
Change-Id: I99a38e2461ee9bd06dfe68758490afe75c2475ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176750
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31476}
Since dependencies on Abseil need to be statically linked in case
Chromium is built with is_component_build=true, this CL introduces a new
parameter for C++ library rtc_* templates (rtc_library, rtc_source_set
and rtc_static_library). This parameter (called "absl_deps") will result
in a dependency on the Abseil component (//third_party/abseil-cpp:absl)
when is_component_build=true or on the normal granular Abseil target
when is_component_build=false.
Bug: chromium:1046390
Change-Id: Iddca886926a7874488701bc9d79affb00cca72d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176447
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31441}
This reverts commit 6958d2c6f0.
Disable the test on iOS.
Bug: None
Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31437}
According to gmock guidelines, mocks for API classes should live
in the same package which owns the API.
No-Try: True
Bug: webrtc:11642
Change-Id: Ib105a1806cc710bc4cff752b8950e981bb4bc326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176381
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31429}
This is a prerequisite to implementing it externally.
Bug: webrtc:11525
Change-Id: I9cb3b4418396485d3eb9f25cafa51cbff6db7817
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176401
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31418}
To support multiple participants video quality analyzer may need to know
peer names in advance to simplify internal structures and metrics
reporting.
Bug: webrtc:11631
Change-Id: I4ffb1554ab7f0e015b8e937b7ffddd55aba9826f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176364
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31415}
Add peer name to video quality analyzer interface to make it possible to
add multipeer support.
Change-Id: I2570cd4481503c8634bdd91208b3dd2fa1d62029
Bug: webrtc:11631
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176329
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31395}
This should help debugging when adaptation is or is not happening
unexpectedly. Log spam is prevented by not logging if the same
result happened to the same resource already and we haven't
adapted since then.
Bug: webrtc:11616
Change-Id: Ia6c5cc35061d252f1c66f2f2bf3b94d2485498d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176221
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31378}
This reduces the number of threads allocated per PeerConnection when
more than one PC is needed.
Bug: webrtc:11598
Change-Id: I3c1fd71705f90c4b4bbb1bc3f0f659c94016e69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31347}
This change adds an optional delay to NetEq's output. Note, this is not
equivalent to increasing the jitter buffer with the same extra length.
Bug: b/156734419
Change-Id: I8b70b6b3bffcfd3da296ccf29853864baa03d6bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175110
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31343}
This CL extends the WebRTC testing API to allow audioproc_f -based
testing using a pre-created AudioProcessing object. This is an
important feature to allow testing any AudioProcessing objects
that are injected into WebRTC.
Beyond adding this, the CL also changes the simulation code to
operate on a scoped_refptr<AudioProcessing> object instead of a
std::unique<AudioProcessing> object
Bug: webrtc:5298
Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31319}
This reverts commit 576ad5d510.
Reason for revert: Causes compile error in Chrome.
Original change's description:
> Make TransformableVideoFrameInterface::GetMetadata pure virtual.
>
> GetMetadata() has been implemented downstream and can be made pure
> virtual.
>
> Bug: chromium:1069295
> Change-Id: I62a3be6106552d2d82d8c413c6f523d31626b0d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175001
> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31281}
TBR=hta@webrtc.org,marinaciocea@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1069295
Change-Id: I5915270d5b8dab9fc30a07f22fddedb29beca01a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175620
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31304}
This adds priority to the API configuration of datachannels,
and passes the value in the OPEN message.
It does not yet influence SCTP prioritization of messages.
Bug: chromium:1083227
Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31287}
GetMetadata() has been implemented downstream and can be made pure
virtual.
Bug: chromium:1069295
Change-Id: I62a3be6106552d2d82d8c413c6f523d31626b0d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175001
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31281}
This reverts commit 28685dc08c.
Reason for revert: Speculative reland after looking into downstream
failures. It's possible that carryover state from unrelated tests
running in parallel was causing failures.
Original change's description:
> Revert "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase."
>
> This reverts commit 46b3bc6c24.
>
> Reason for revert: Speculative revert. Breaks downstream project
>
> Original change's description:
> > Make sure that "current" rtc::Thread instances are always current for TaskQueueBase.
> >
> > This is a necessary part of fulfilling the TaskQueueBase
> > interface. If a thread does not register as the current TQ, yet offers
> > the TQ interface, TQ 'current' checks will not work as expected and
> > code that relies them (TaskQueueBase::Current() and IsCurrent())
> > will run in unexpected ways.
> >
> > Bug: webrtc:11572
> > Change-Id: Iab747bc474e74e6ce4f9e914cfd5b0578b19d19c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175080
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31254}
>
> TBR=mbonadei@webrtc.org,tommi@webrtc.org
>
> Change-Id: I69ff3355f0ec447b25604bd95fdacbdb4d4f3f27
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175104
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31259}
TBR=mbonadei@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org
# Not skipping CQ checks because this is a reland.
Bug: webrtc:11572
Change-Id: I00c82d99af8e05851769e09cb682b5b73895a6f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175133
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31273}
Define MockTransformableVideoFrame in api/test to be use in Blink
tests.
Bug: chromium:1069295
Change-Id: I960d8228eddbdf8263b731814a00fd02d0a6c54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175136
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31272}
Several tests leave pending tasks behind after executing, which may
affect the state of subsequent tests. This CL isolates each test in
the sense that a dedicated Thread instance is created per test and
then pending tasks are flushed and the Thread instance deleted.
Down the line we may want to improve on this and flag those tests
that leave pending tasks/timers etc.
Change-Id: Ibaf3719a9974c57ac2169edca0e2a06a9ea6c78f
Bug: webrtc:11574
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175132
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31268}
Define VideoHeaderMetadata, containing a subset of the metadata in RTP
video header, and expose it the TransformableVideoFrameInterface, to
enable web application to compute additional data according to their own
logic, and eventually remove GetAdditionalData() from the interface.
Bug: chromium:1069295
Change-Id: Id85b494a72cfd8bdd4c0614844b9f0ffae98c956
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174822
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31265}
This reverts commit 46b3bc6c24.
Reason for revert: Speculative revert. Breaks downstream project
Original change's description:
> Make sure that "current" rtc::Thread instances are always current for TaskQueueBase.
>
> This is a necessary part of fulfilling the TaskQueueBase
> interface. If a thread does not register as the current TQ, yet offers
> the TQ interface, TQ 'current' checks will not work as expected and
> code that relies them (TaskQueueBase::Current() and IsCurrent())
> will run in unexpected ways.
>
> Bug: webrtc:11572
> Change-Id: Iab747bc474e74e6ce4f9e914cfd5b0578b19d19c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175080
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31254}
TBR=mbonadei@webrtc.org,tommi@webrtc.org
Change-Id: I69ff3355f0ec447b25604bd95fdacbdb4d4f3f27
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175104
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31259}
This is a necessary part of fulfilling the TaskQueueBase
interface. If a thread does not register as the current TQ, yet offers
the TQ interface, TQ 'current' checks will not work as expected and
code that relies them (TaskQueueBase::Current() and IsCurrent())
will run in unexpected ways.
Bug: webrtc:11572
Change-Id: Iab747bc474e74e6ce4f9e914cfd5b0578b19d19c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31254}
This allows users to inject the residual echo detector, as a step toward making it an optional part of compilation.
Bug: webrtc:11292, webrtc:11539
Change-Id: I7fcc8dbaced67a82851cd6cdcbc115eb01c21fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174040
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31222}
Add peer's name to params and use it for logging and metrics naming
for whole peer related metrics.
Bug: webrtc:11479
Change-Id: Ia7e3fc4839c90a958d66910614515ac02a96e389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174752
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31215}
Modernise functions to unified MOCK_METHOD macro,
delete few deprecated functions on the way.
add one missing function (in MockEncodedImageCallback)
Remove proxy mock function (in MockVideoBitrateAllocatorFactory)
Remove default constructors and destructors
Bug: None
Change-Id: Ibebb0d9e3c9be5877649af7bde8b87222ddf04fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174751
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31195}
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.
Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
This interface has a couple of issues. Primarily for me, it makes it
difficult work with the paced sender as we need to either temporarily
release a lock or force a thread-handover in order to avoid a cyclic
lock order.
For video in particular, its behavior is also falky since header sizes
can vary not only form frame to frame, but from packet to packet within
a frame (e.g. TimingInfo extension is only on the last packet, if set).
On bitrate allocation, the last reported value is picked, leading to
timing issues affecting the bitrate set.
This CL removes the callback interface and instead we simply poll the
RTP module for a packet overhead. This consists of an expected overhead
based on which non-volatile header extensions are registered (so for
instance AbsoluteCaptureTime is disregarded since it's only populated
once per second). The overhead estimation is a little less accurate but
instead simpler and deterministic.
Bug: webrtc:10809
Change-Id: I2c3d3fcca6ad35704c4c1b6b9e0a39227aada1ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173704
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31185}
After the migration to passing frame video source implementation directly, part of the peer connection framework code became redundant. Removing screen_share_config and capturing_device_index from the VideoConfig is to be done in later reviews.
Bug: webrtc:11534
Change-Id: I7a8ea85d26d00fb5bfe7ec0d2facef9c44a0f749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174541
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31178}
This reverts commit 9a925c9ce3.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839d.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
This is one last CL that includes the rest of VoIP API implementation.
Bug: webrtc:11251
Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31168}
In order for WebRTC to be able to include packet overhead in its
bitrate calculations, the AudioEncoder::GetFrameLengthRange()
function must be implemented by all audio encoders. Making this
member function pure virtual as per the following PSA:
https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0
Bug: webrtc:11427
Change-Id: I30d297ef05f57453bfc257624729559057cad118
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31127}
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}
Add new methods in the FrameTransformerInterfaces, passing the ssrc on
registering the transformed frame callback, to associate separate frame
transformer sinks for each ssrc. Same for unregister.
Bug: chromium:1065838
Change-Id: I8a406815e9d0cce5199f9df06c286d8b10d75b4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173183
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31047}
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 reverts commit 8e8b36a94a.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
If a STUN attribute is in the "comprehension-required" range
(0x0000-0x7FFF), and the implementation does not recognize it, this
should be treated as an error (as per RFC5389), with different behavior
depending on the type of the message received.
Bug: webrtc:9063
Change-Id: Ic31b0cdd3c26772c21d770b44fe4ee4a1b47030a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/64500
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30925}
To allow construct that object from an existent string_view without explicit conversion
Bug: webrtc:11428
Change-Id: I38d93573be72e307bdf7068a6300d10cf46d2d62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171689
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30904}
To migrate on new GetStats API and properly support target encode bitrate
for regular, simulcast and svc cases we need to calculate it inside video
quality analyzer getting values from SetRates in VideoEncoder.
Bug: webrtc:11381
Change-Id: Ia37acac764ed3c30f64cdbfda8906d543fa03ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171501
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30881}
This CL renames the shadow filter in AEC3 to have the more accurate name
coarse filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "shadow" with "coarse" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: I28d6041d0d34e85f8f8048d004b44a1a5f07bb07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170981
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30846}
This CL replaces a few usages of const std::string& with
absl::string_view, to comply closer with
https://abseil.io/tips/1.
Bug: webrtc:11428
Change-Id: Ibf6fac9b084cb21e17db63f73d667793ab9cafeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170466
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30845}
This CL renames the main filter in AEC3 to have the more accurate name
refined filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "main" with "refined" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: Ifd0aab34e291736a2250e0986348404618630b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170825
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30843}
Track the number of samples that are decoded until a fallback to
software decoder happens.
Bug: chromium:1061376
Change-Id: Ida3ae94034ec83a6d28001cb7be343b8b99b99c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170468
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30814}
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
I'll hold on to the root OWNER for a bit longer for convenience.
Bug: None
Change-Id: I13303ba726fed612adc74008eeaaeadf9595e084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170047
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30727}
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 does not add it in default SDP offer.
Bug: webrtc:10739
Change-Id: I4e73f4497989fc34f3676927921a4dabb5926096
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169729
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30707}
The decision to route audio packets to a separate overuse detector
is off by default and requires the field trial
WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/
The parameters control the threshold for switching over to the
audio overuse detector if we stop receiving feedback for video.
Bug: webrtc:10932
Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30694}
The degradation preference is now based on the content hint of the track
if it's unspecified.
Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30691}
It can only be one of four possible values, so it never made sense
for it to be a double. Other than the fact that its neighbor
bitrate_priority is a double, and they're both defined as the same enum
in the web spec. However, while bitrate_priority being a double
offers more flexibility than the web spec, network_priority being a
double is only confusing.
Bug: webrtc:5658
Change-Id: I0784c116f3260c4b3a8b99a3cd85c8d66017e46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30685}
This avoid duplication. As part of this moving the overhead calculation
to the IP address class so it's easier to find and more natural to use.
Bug: webrtc:9883
Change-Id: If4d865f445bc1a302572896932966ce30294e339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169445
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30657}
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}
Previously, only lvalues were tested, and only lvalues worked.
Bug: webrtc:11389
Change-Id: I524e9d63e0840c3ba274dbe2062d78f72d79019d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169347
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30644}
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}
This matches the web API more, since the equivalent type there is named
RTCPriorityType.
Bug: webrtc:5658
Change-Id: I301fed8319f7e582b558fe7cd0deee1290708c4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30613}
Specifically api, pc and p2p.
Bug: None
Change-Id: I2ba19aaac5ca11a5282593f0db06bba326fe6891
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30609}
This is a reland of 1a290e4495
after fixing the downstream projects.
Original change's description:
> Remove old-style OnFailure callbacks
>
> Also delete default implementation of new-style OnFailure,
> since it can't call the deprecated function.
>
> Deprecating the old-style OnFailure callback turned out to be impossible,
> since one can't have the new-style callback call the old-style one.
>
> Bug: chromium:589455
> Change-Id: Icf529ddb02d99ad9e205095d5a1fbeb0da91dd0e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146219
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30570}
Bug: chromium:589455
Change-Id: I7227e3c6886c504043b019b621e45658cbd6fd53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168941
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30583}
Also delete default implementation of new-style OnFailure,
since it can't call the deprecated function.
Deprecating the old-style OnFailure callback turned out to be impossible,
since one can't have the new-style callback call the old-style one.
Bug: chromium:589455
Change-Id: Icf529ddb02d99ad9e205095d5a1fbeb0da91dd0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146219
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30570}
After chromium switches to using these, they'll be changed to an enum.
Bug: webrtc:5658
Change-Id: Ic5d7d4651d204c31822194bd02c587e5b887ee17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168562
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30551}
It is now possible to set the target bitrate for iSAC for the fixed
point implementation. Unit tests added.
Bug: webrtc:11360
Change-Id: I60225d4ca1363cdacf18931e7cf412c5aec8d8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168529
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30526}
Change RtpCodecCapability::parameters and RtpCodecParameters::parameters
to map from unordered_map to get welldefined FMTP lines.
Bug: webrtc:7061
Change-Id: Ie61f76bbab915d72369e36e3f40ea11838827940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168190
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30512}
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}
If the incoming frame is a native frame but the native encoder fails,
we should ensure the fallback encoder can handle the native frame. If
not then the native frame should be scaled and converted.
Bug: webrtc:11346
Change-Id: I692350dc69b5ce2db7ba5ee98d28f94cb12054cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168345
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30504}
No-Try: True
Bug: None
Change-Id: I632d5384321c88202a23cc3fa6938afac0f796ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168460
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30494}
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}
This TODO says this metric is only available for audio and should also
be implemented for video, but ever since M76 this has been implemented
for both audio and video (https://crbug.com/webrtc/10450).
TBR=guido@webrtc.org, hta@webrtc.org
NOTRY=True
Bug: webrtc:10450
Change-Id: Icf2b60fdacae606c66f9d03492f107df9e32ba33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168343
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30485}
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}
These factories suppose to replace set of old constexpr factories that
takes parameter as template rather than function parameter,
as well as fix function naming to follow style guide of the second set
of factory functions.
Bug: None
Change-Id: Icd76302b821b2a4027f9d6765cf91bc9190f551c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167521
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30482}
This function is obsolete now that config-based functions are available.
The command-line parsing should not happen here but in the executable
that uses these functions.
Bug: webrtc:11005
Change-Id: I618d12503123e3e1fd6e572a045372c622043a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30421}
Also adding kHz factory function for Frequency class.
Bug: webrtc:9883
Change-Id: Ide44910d50eb9616de2bb0c66b8c62493d2be92e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167725
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30416}
Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport. This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).
This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.
Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
Since RTC_DCHECK was made constexpr compatible, we can now
make the unit classes fully constexpr.
Bug: webrtc:9883
Change-Id: I18973c2f318449869cf0bd45699c41be53fba806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30403}
This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
When creating a NetEqController it can be useful to have access to a
webrtc::Clock*. Also, NetEqControllers should have access to the
contents of the sync buffer when making decisions.
Bug: webrtc:11005
Change-Id: I7fdba75ce661b2ace52458620a8c1f3c990e5ac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167208
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30368}
This reverts commit 133bf2bd28.
Reason for revert: Breaks Chromium import due to flaky test in Chromium.
Original change's description:
> Reland "Distinguish between send and receive codecs"
>
> This reverts commit e57b266a20.
>
> Reason for revert: Fixed negotiation of send-only clients.
>
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762.
> >
> > Reason for revert: breaks negotiation with send-only clients
> >
> > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> >
> > Original change's description:
> > > Distinguish between send and receive codecs
> > >
> > > Even though send and receive codecs may be the same, they might have
> > > different support in HW. Distinguish between send and receive codecs
> > > to be able to keep track of which codecs have HW support.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30284}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30292}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30348}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30360}
This commit enables developers to configure the "," delay value from
the WebRTC spec value of 2 seconds. This flexibility allows developers
to comply with existing WebRTC clients.
Bug: webrtc:11273
Change-Id: Ia94b99e041df882e2396d0926a8f4188afe55885
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165700
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30354}
This reverts commit e57b266a20.
Reason for revert: Fixed negotiation of send-only clients.
Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30292}
TBR=steveanton@webrtc.org,kron@webrtc.org
Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30348}
This makes it easier to maintain consistency between real time
and simulated time modes.
The RealTimeController is updated to use an explicit main thread,
this ensures that pending destruction tasks are run as the network
emulator goes out of scope.
Bug: webrtc:11255
Change-Id: Ie73ab778c78a68d7c58c0f857f14a8d8ac027c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30342}
This change clarifies the semantics of this field:
unset: Depends on context.
== 0: Invalid.
== 1: No temporal layering.
>= 2: Temporal layering.
We should try to remove the wrapping optional later.
Bug: webrtc:11297
Change-Id: Id765f2dc1d31a4ba3cd424978ac6054cd60152ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166528
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30336}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This squashes together several input signals that were spread out
through several calls into a single method and calling place:
SetEncoderSettings(), invoked from ReconfigureEncoder(). This is added
to the abstract interface.
This makes the following methods obsolete which are removed:
- SetEncoder(): The VideoEncoder was only used for GetEncoderInfo();
the VideoEncoder::EncoderInfo is now part of the EncoderSettings.
- SetEncoderConfig(): The VideoEncoderConfig is part of
EncoderSettings. The config is used for its codec_type and
content_type enums.
- SetCodecMaxFrameRate(): The max frame rate was the same as
VideoCodec::maxFramerate. VideoCodec is now part of EncoderSettings.
There may be some overlap in information between EncoderConfig and
VideoCodec, but that is outside the scope of this CL, which only makes
sure to bundle encoder settings-like information into one input signal.
Bug: webrtc:11222
Change-Id: I67c49c49c0a859cb7d5051939a461593c695a789
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166602
Reviewed-by: Erik Språng <sprang@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@{#30332}
The extension (and thus structures to carry it) are designed
in particular for client<->SFU link. Putting the structure into api
acknowledges it can be reused by SFU projects
Bug: webrtc:10342
Change-Id: I8ca1f5046abadf6aa16200443c4892e9a2a928b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166467
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30324}
This is a reland of 2a11b2451a
There are no changes compared to the first attempt.
Original change's description:
> Enable using a custom NetEqFactory in simulations
>
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}
TBR=kwiberg
Bug: webrtc:11005
Change-Id: I4aa377e05916bd23f8f63aece9d0e27731c80d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166465
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30319}
These constants describes how value should be put on the wire and thus
belong to the extension builder/writer class rather than extension value class
Bug: None
Change-Id: I65ca3923eddcc2e48563ad69b98356c159ad86be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30305}
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.
Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
This is a reland of b70c5c5ce9
Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
>
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}
Bug: webrtc:11255
Change-Id: If65cd56b59158cebec5609407a721fbdb47cfd1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30294}
This reverts commit c0f25cf762.
Reason for revert: breaks negotiation with send-only clients
(webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
(peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
(peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30292}
Even though send and receive codecs may be the same, they might have
different support in HW. Distinguish between send and receive codecs
to be able to keep track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30284}
This patch improves handshake wrt GOOG_PING support so that
- if goog_ping_enable: sender send it's goog-ping version until it gets
STUN_BINDING_RESPONSE
- receiver only sends it's goog-ping-version if getting a
goog-ping-version in the request
This means that the overhead of STUN_ATTR_GOOG_MISC_INFO is only
- added on STUN_BINDING_REQUEST until a response is received.
- added on STUN_BINDING_RESPONSE if remote peer request it.
This is wire compatible with older versions so that
- new sender will enable GOOG_PING with new/old receiver.
- old sender will enable GOOG_PING with old receiver.
- old version will not enable GOOG_PING with new receiver
(receiver expecting sender to announce first).
BUG: webrtc:11100
Change-Id: Ib3434c593988188150f4c7506918139aaf138d0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165787
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30269}
This reverts commit b0e0728159.
Reason for revert:
Causes Chromium tests to timeout, preventing rolls into Chromium.
Original change's description:
> Replaces SynchronousMethodCall with rtc::Thread::Invoke.
>
> Given that we already have Thread:.Invoke that can be used with lambda,
> SynchronousMethodCall doesn't add any value.
>
> This simplification prepares for simulated time peer connection tests.
>
> Bug: webrtc:11255
> Change-Id: I478a11f15e30e009dae4a3fee2120f6d7a03355f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165683
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30217}
TBR=steveanton@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11255
Change-Id: I9d3aa218013129db7a09a77500a0547ce9ae341a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166047
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30265}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Given that we already have Thread:.Invoke that can be used with lambda,
SynchronousMethodCall doesn't add any value.
This simplification prepares for simulated time peer connection tests.
Bug: webrtc:11255
Change-Id: I478a11f15e30e009dae4a3fee2120f6d7a03355f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165683
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30217}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
This CL also removes RTCRtpStreamStats::associateStatsId, which is the
legacy name for this stat, which was never implemented (existed in C++
but the member always had the value undefined and was thus never exposed
in JavaScript).
Bug: webrtc:11228
Change-Id: I28c332e4bdf2f55caaedf993482dca58b6b8b9a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30171}
The IPAddress class (32 bytes) was copied for each invocation.
This CL also saves some bytes in generated binary.
Bug: webrtc:9855
Change-Id: I40f2fe8570ee30d1d2251fddd56131ca4c3e7155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164521
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30147}
This CL ensures that the no data vectors in the reverb computer code
are fixed. This allows arbitrary long filters to be used, and ensures
that a minimum required heap size is used.
Bug: webrtc:8671
Change-Id: I7085ed262a3f5965d796270434b6578f4030606e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162661
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30115}
This CL propagates a "closed" signal from DTLS up to the
SCTP section of the data channel controller, where it causes
closing of all open datachannels.
Bug: chromium:1030631, webrtc:10360
Change-Id: I88bb9e1aff5c25f330edfd092ef609d4fcc3a9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162206
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30099}
This reverts commit f22af3cca7.
Reason for revert: Downstream tests have been updated.
Original change's description:
> Revert "Distinguish between send and receive video codecs"
>
> This reverts commit 18314bd8d2.
>
> Reason for revert: Breaks downstream test.
>
> Original change's description:
> > Distinguish between send and receive video codecs
> >
> > Even though send and receive codecs are the same,
> > they might have different support in HW.
> > Distinguish between send and receive codecs to be able to keep
> > track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30041}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30042}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30078}
This is used to allow using a pre-configured builders as arguments to
fixture code.
Bug: webrtc:9510
Change-Id: I7837d284580fdbc926535ce5b2d8f582056534ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161948
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30070}
ScalingSettings has some public constructors. These should be
able to be called from exteranl code. However, a linker fails
on windows because ScalingSettings doesn't have RTC_EXPORT.
Bug: chromium:1031965
Test: build crrev.com/c/1949841
Change-Id: Iddaea77f87c52edbe8f77551322d7aa198bc0aeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30063}
This patch introduces a new type of STUN ping,
GOOG_PING_REQUEST/RESPONSE which is similar
to a STUN_BINDING but does not transmit any values.
The Connection class automatically sends these if
no STUN attributes has changed since last call to Connection::Ping()
if the remote peer has signaled that it supports it.
BUG=webrtc:11100
Change-Id: Ib1b590f0b90ca6cb56f2eb07cd62f976e246bc8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159961
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30062}
This reverts commit 18314bd8d2.
Reason for revert: Breaks downstream test.
Original change's description:
> Distinguish between send and receive video codecs
>
> Even though send and receive codecs are the same,
> they might have different support in HW.
> Distinguish between send and receive codecs to be able to keep
> track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30041}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30042}
Even though send and receive codecs are the same,
they might have different support in HW.
Distinguish between send and receive codecs to be able to keep
track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30041}
This is the start of generating compliant errors, including diagnostics,
when datachannels close because of errors.
Bug: chromium:1030631
Change-Id: I39aa41728efb25bca6193a782db4cbdaad8e0dc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161304
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30034}
Introduce FrameGeneratorInterface to make FrameGenerator API available
for downstream projects.
Bug: webrtc:10138
Change-Id: I4216775e4b8b54c3f1c72d67ffbda31eb082fd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161234
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30009}
This makes the class easier to use at a minor cost of making it slightly
more magic.
Bug: webrtc:9883
Change-Id: If807cfbf046615333c3bcd3b58a001813102a9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30008}
This change finally wires up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.
Bug: chromium:1013590
Change-Id: I76f8226752294aee8fe137d1a78ee66548900cc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161095
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30003}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
When building certain Chromecast build flavors in component build mode,
there are some link errors due to symbols not being exported. This CL
fixes those issues.
TBR: kwiberg@webrtc.org
Bug: None
Change-Id: I408f0a84b8ac4610cc6b5aa6ff58248ea82c9c66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161148
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29992}
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.
Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames
can flow to sinks installed in VideoTrackSourceInterface.
Bug: chromium:1013590
Change-Id: I0779932c251a2159880a39b2d42d5ce439cc88e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161090
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29988}
It should fix compilation errors that happen on some iOS bots saying
"definition of implicit copy assignment operator for 'Foo'
is deprecated because it has a user-declared copy constructor"
Bug: webrtc:11162
Change-Id: Ife3d1a800ed6a4cd08bdfd156cd0e320504ee8dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161221
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29984}
This CL adds code for doing signal-dependent downmixing
before the delay estimation in the multichannel case.
As part of the CL, the unittests of the render delay
controller are corrected. However, as that caused some of
them to fail, the CL (for now) as well disables the failing
test.
Bug: webrtc:11153,chromium:1029740, webrtc:11161
Change-Id: I0b765c28fa5e547aabd6dfbd24b626ff9a16346f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161045
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29980}
This CL decouples NetEqFactory and AudioDecoderFactory.
AudioDecoderFactory is used in more places than just inside of NetEq, so
decoupling these makes sense.
Bug: webrtc:11005
Change-Id: I78dd856e4248e398e69a65816b062ef30555b055
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161005
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29961}
Since this macro can be considered public, it makes sense to prefix it
with WEBRTC_ (also to avoid potential conflicts with client code).
This CL also removes some definitions of this macro in order to define
it only where it is strictly needed (it is only used in a .cc file).
Bug: webrtc:11142
Change-Id: Idce7389301e71d8434e238b3cf4ceaa9cf97cd87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161008
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29957}
This CL changes the downmixing of the input to the delay estimation
for surround/stereo signals to be off by default.
A kill-switch is also added for enforcing the downmix to be on.
Bug: webrtc:10913
Change-Id: I1030fef593ba56416deeb13b80d2f3812bffb9ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161012
Commit-Queue: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29951}
This patch introduces 3 new functions on StunMessages
- Clone, copy a message
- IsStunMethod, verifies that a buffer is a StunMessage
w/o requring a fingerprint
- EqualAttributes, compare attributes in two stun messages
(with filter)
This methods will be used to implement GOOG_PING
BUG=webrtc:11100
Change-Id: I284726c74aa0437be0bb9fbcf943c7d64a18acec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160281
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29950}
There are edge cases where the caching of encoder info will cause
issues. For instance if a sub-encoder fails en Encode call and falls
back to some other implementation, or if the fps targets shift due to
SetRates() triggering new layers to be enabled.
This CL forces a complete rebuild on every call to GetEncoderInfo().
It also adds new logging of when the info changes, as debugging issues
can be very time consuming if we can't tell that happened.
Bug: webrtc:11000
Change-Id: I7ec7962a589ccba0e188e60a11f851c9de874fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29938}
Use that conversion instead of duplicating it in call/
Bug: webrtc:11042
Change-Id: I035b161d429ec339dd2ad9e9ed3ede5045fb6199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160881
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29936}
This reverts commit 15be5282e9.
Reason for revert: crbug.com/1028937
Original change's description:
> Add support for RtpEncodingParameters::max_framerate
>
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}
TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
Many WebRTC users need only Opus, and no other audio codecs. This
makes it convenient for them to do the right thing.
To prove that the new factories work, use them in
PeerConnectionEndToEndTest.
Bug: webrtc:11130
Change-Id: I2c2450ba0fb33ef3b50da8f6cd325cad6b1e59a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160648
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29921}
The NetEqFactory is currently expected to wrap the AudioDecoderFactory,
but this turns out not to be a good idea. Instead, it makes more sense
to pass the AudioDecoderFactory through the CreateNetEq method.
Bug: webrtc:11005
Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29918}
This change implements the methods in VideoTrackSourceInterface
that are related to encoded output.
Bug: chromium:1013590
Change-Id: Id9ddbc00a7098e9b44cee1517c69002865a5fb33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159926
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29912}
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.
Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
Implements an alternative to the dominant nearend detector.
Bug: b/130016532
Change-Id: If4867d58aad036ccf4e456ef81689b8db0284f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159865
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29906}
Creates an abstraction for an "alarm clock" which can schedule
time-controller callbacks and exposes a time controller driven by
an external alarm.
Bug: webrtc:9719
Change-Id: I08c2aa9dba25603043bfba48f55c925716a55bae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158969
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29879}
All downstream users have been moved to the new one.
Bug: webrtc:11091
Change-Id: Ia18d0df94a7b95b1a58b4a53cfb195c61ef59ffd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160201
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29873}
This patch adds
- Attribute: STUN_ATTR_GOOG_MESSAGE_INTEGRITY_32
which is a ordinary message integrity but truncated to 32-bit
- Method: GOOG_PING,
which will be used for webrtc:11100
Both the attribute and the method has been registered at iana,
https://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml#stun-parameters-4
BUG=webrtc:11100
Change-Id: Iddd5614473fd6f18fbbe76e72d047c617df7123f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160180
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29864}
This reverts commit e6eded31e6.
Reason for revert: A better method for communicating encoded frames in VideoTrackSourceInterface surfaced.
Original change's description:
> VideoFrame: Store a reference to an encoded frame
>
> Enable webrtc::VideoFrame to store a reference to an encoded frame.
>
> Bug: chromium:1013590
> Change-Id: Id5a06f1c7249f104dfd328f08677cf8001958f0d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158788
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29809}
TBR=ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,handellm@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1013590
Change-Id: I46384b7997e7b1cd3a2a2042cf17890fc977cca3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160204
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29863}
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.
Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
This change adds a new type of sink for consuming encoded data from
a video source.
Bug: chromium:1013590
Change-Id: Ia7c4e372190c3d6bc007a0d4deb05c2d1bce58d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159927
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29856}
This part of the effort to implement A/V sync metric.
Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
Add tests for different UpdateRect methods as they are no longer trivial
This change will enable providing useful update rects after scaling
is done.
Bug: webrtc:11058
Change-Id: I2311dbbbb5eca5cfaf845306674e6890050f80c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159820
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29835}
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).
I checked what our downstream users are actually using, and it's
cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage
I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.
There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.
Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.
Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
This is a reland of 53e157d25c
The issue has been fixed in
https://chromium-review.googlesource.com/c/chromium/src/+/1917204.
Original change's description:
> Force Chromium deps on the WebRTC component.
>
> This CL adds a visibility check to the rtc_* GN templates in order
> to force Chromium to depend only on publicly visible targets from
> //third_party/webrtc_overrides and not from //third_party/webrtc.
>
> This is required in order to ensure that the Chromium's component
> builds continues to work correctly without introducing direct
> dependency paths on WebRTC that would statically link it in multiple
> shared libraries.
>
> Bug: webrtc:9419
> Change-Id: Ib89f4fc571512f99678ee4f61696b316374346d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154344
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@chromium.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29806}
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I7123d1b44ddbc23b11d9fa25aa39aa420359e33d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159922
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29816}
This reverts commit 53e157d25c.
Reason for revert: Breaks Chromium iOS FYI bots.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/5088
Original change's description:
> Force Chromium deps on the WebRTC component.
>
> This CL adds a visibility check to the rtc_* GN templates in order
> to force Chromium to depend only on publicly visible targets from
> //third_party/webrtc_overrides and not from //third_party/webrtc.
>
> This is required in order to ensure that the Chromium's component
> builds continues to work correctly without introducing direct
> dependency paths on WebRTC that would statically link it in multiple
> shared libraries.
>
> Bug: webrtc:9419
> Change-Id: Ib89f4fc571512f99678ee4f61696b316374346d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154344
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@chromium.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29806}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,dpranke@chromium.org
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: Id4d906910d569a3e5db3afef8c03672fba6dad81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29813}
Enable webrtc::VideoFrame to store a reference to an encoded frame.
Bug: chromium:1013590
Change-Id: Id5a06f1c7249f104dfd328f08677cf8001958f0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158788
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29809}
This CL adds a visibility check to the rtc_* GN templates in order
to force Chromium to depend only on publicly visible targets from
//third_party/webrtc_overrides and not from //third_party/webrtc.
This is required in order to ensure that the Chromium's component
builds continues to work correctly without introducing direct
dependency paths on WebRTC that would statically link it in multiple
shared libraries.
Bug: webrtc:9419
Change-Id: Ib89f4fc571512f99678ee4f61696b316374346d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154344
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Dirk Pranke <dpranke@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29806}
This CL enables extracting the linear AEC output,
allowing for more straightforward
testing/development.
Bug: b/140823178
Change-Id: I14f7934008d87066b35500466cb6e6d96f811688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153672
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29789}
This CL adds a dependecy on rtc_base/system:rtc_export to rtc_event but
only when built as part of Chromium (since rtc::Event should not be
used outside of WebRTC).
It also adds other missing RTC_EXPORTS.
Bug: webrtc:9419
Change-Id: Ib338004a5404a6b3c7929e146c29ad42572632cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159692
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29781}
Add ability to provide custom implementation of rtc::VideoSourceInterface
as source for video track in PC-framework based media quality tests.
Bug: webrtc:10138
Change-Id: I8ffd3015230c733a0a9a2e97fd4bb93a0c02b283
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159680
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29776}
to align with chromium scoped_refptr implementation
and prefer move over copy in some cases.
Bug: webrtc:11078
Change-Id: I3178e74e611e4b23435668878e6bcc98bc2ce77d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159541
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29768}
This is the same as the existing version, except it uses the Operations
Chain. As such, if an asynchronous operation that uses the chain is
currently pending, such as CreateOffer() or CreateAnswer(),
AddIceCandidate() will not happen until the previous operation
completes.
Bug: chromium:1019222
Change-Id: Ie6e5fc386fa9c29b5e2f8e3f65bfbaf9837d351c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158741
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29704}
Starting from [1], explicit template declaration/definition is in use
for this template so there is no need to RTC_EXPORT its declaration.
Doing so leads to this error on clang-cl:
../../third_party/webrtc\api/stats/rtc_stats.h(372,1): error: explicit instantiation declaration should not be 'dllexport' [-Werror,-Wdllexport-explicit-instantiation-decl]
WEBRTC_DECLARE_RTCSTATSMEMBER(bool);
^
../../third_party/webrtc\api/stats/rtc_stats.h(369,3): note: expanded from macro 'WEBRTC_DECLARE_RTCSTATSMEMBER'
extern template class RTC_EXPORT_TEMPLATE_DECLARE(RTC_EXPORT) \
^
../../third_party/webrtc\api/stats/rtc_stats.h(287,7): note: attribute is here
class RTC_EXPORT RTCStatsMember : public RTCStatsMemberInterface {
^
../..\third_party/webrtc/rtc_base/system/rtc_export.h(24,31): note: expanded from macro 'RTC_EXPORT'
Full log: https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/430931
[1] - https://webrtc-review.googlesource.com/c/src/+/158795
Bug: webrtc:9419
Change-Id: I9f0893ae26b45049f186e19f862a1d138a320a24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158891
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29703}
This is a reland of 9dda1b3a48
Original change's description:
> Correct AEC3 multichannel functionality activation
>
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
>
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}
Bug: webrtc:10913
Change-Id: I1cb3d0de61ea0b299158ca85433f2442c65c196f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158886
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29690}
These two annotations are now needed to correctly compile Chromium
with is_component_build=true and the WebRTC component.
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: Id5603cf747357c0c2a4b41684eb4fd607cccfdea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29686}
This is a reland of 9dda1b3a48
Original change's description:
> Correct AEC3 multichannel functionality activation
>
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
>
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}
Bug: webrtc:10913
Change-Id: Ibfe4e8a51183390a4054514bb294c89c2ea201e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158880
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29685}
This CL works around an "Explicit specialization after instantiation
error" when building with clang-cl and is_component_build=true (see
crbug.com/1018579). On top of that it uses "template instantiation
declarations/declarations" in order to avoid to instantiate the
template in clients code.
TBR: hbos@webrtc.org
Bug: webrtc:9419, chromium:1018579
Change-Id: I1b2862de678586afc81e8f7a407947322f8a06c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158795
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29683}
This reverts commit 9dda1b3a48.
Reason for revert: The CL is causing downstream issues
Original change's description:
> Correct AEC3 multichannel functionality activation
>
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
>
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: Ic487f77f5c11485a0f25a2a1d3797d0ec956f913
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158797
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29678}
This CL corrects the AEC3 multichannel activation
to also work for the case when a factory is used
for the activation.
Bug: webrtc:10913
Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29676}
Injecting both a custom NetEqFactory and an AudioDecoderFactory is not
supported, in that case the AudioDecoderFactory should be wrapped inside
the NetEqFactory.
Bug: webrtc:11005
Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29673}
Move the GCM srtp cipher suites below the default SRTP_AES128_CM_SHA1_80 one.
This will not negotiate them by default since they have an impact on packet overhead for audio-only calls.
GCM can still be negotiated if the peer offers it as preferred cipher suite or answers with just that cipher suite.
BUG=chromium:713701
Change-Id: I79bd4ab827e5c7f55f5550d14db3f4217a7eff86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158404
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29672}
This CL also introduces NetEqFactory and NetEqControllerFactory
interfaces, as well as several convenience classes for working with
them: DefaultNetEqFactory, DefaultNetEqControllerFactory and
CustomNetEqFactory.
Bug: webrtc:11005
Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29671}
This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.
This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.
Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
For the automatic content type detection we need to know if the update
rect is trusted or just not available.
Currently we only care if it's not empty, so in case of no update rect
available, full frame resolution was set as a changed region.
This CL makes the update_rect field optional but should be a no-op in the
current code, as absence of update_rect is treated as a full update via
a new getter method |update_rect_or_full_frame()|.
Bug: webrtc:11058
Change-Id: I913545b71ac2fc845861549ac34eb1b630012109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158673
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29654}
Bug: chromium:1018077
Change-Id: I585d4064f39e5f9d268b408ebf6ae13a056c778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158403
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29628}
This does not help the attached bugs, but it does allow greater control
over what JSON code is running where. Long-term, the JSON library used
for parsing configuration should likely be a library already present
in Chromium builds, to avoid duplication. And if that happens, then
WebRTC bug 9804 may be passé.
Note that this CL also sorts our poisons alphabetically.
Bug: chromium:895814, webrtc:9804
Change-Id: I70c3efe05a0eba9212895407f73978d8216df920
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158400
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29615}
Chromium tests depend on api/task_queue:task_queue_test but it
cannot be added to the WebRTC component in Chromium (which is not
testonly).
A possible solution is to make api/task_queue:task_queue_test
depend on the WebRTC component which lives in Chromium only
when `build_with_chromium=true`.
Bug: webrtc:9419
Change-Id: I1cbe1fb97f21ef7a201d09d2f0f729104d01ed90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157427
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29578}
Type mismatches will silently fail and skip reading a parameter
in the JSON parsing, except when parsing a size_t from a negative int.
This CL updates the parsing to silently ignore negative values provided
for size_t config parameters, instead of explicitly DCHECKing.
Tested: Ran the fuzzer on the crash test case with + without this fix.
Bug: chromium:1016139
Change-Id: I3899e81e1183aa54b708030efeb6e0006b8cd881
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157894
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29568}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
This is a reland of 16d4c4d4fb after
downstream project was updated to be prepared for the new SdpType.
Original change's description:
> Implement rollback for setRemoteDescription
>
> Bug: chromium:980875
> Change-Id: I4575e9ad1902a20937f9812f49edee2a2441f76d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153525
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29422}
TBR=steveanton@webrtc.org
Bug: chromium:980875
Change-Id: Iba8d25bf2dc481b25a03eeae9818bd5f4c3eaa2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156569
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29460}
This CL adds support for multiple channels in the reverb
modelling. As a side effect, it also partly adds multi-channel
supports for the sections of the code.
Beyond adding the multi-channel support, a bug is fixed as part of
this CL. Since the bug fix affects the bitexactness, as a safety
precaution the CL includes the ability to override the bugfix.
Apart from the contributions from the bugfix, the changes have
been verified to be bitexact for a large set of mono recordings.
Bug: webrtc:10913
Change-Id: I1f307b532be85ef4182f8db41384f44d40a25219
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156382
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29456}
This change keeps the original 48 kHz signal and uses it for the
fullband processing given that the following requirements are
fulfilled:
- Input signal is 48 kHz
- Output signal is 48 kHz
- Multiband processing is performed at 32 kHz
- The multiband processing does not modify the original signal
This avoids unnecessary, lossy resampling and band merging.
Bug: b/130016532
Change-Id: I690c26faba07eab0cbff6c0a95a81d89255dd1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155966
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29425}
Bug: chromium:980875
Change-Id: I4575e9ad1902a20937f9812f49edee2a2441f76d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153525
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29422}
Just fixing some minor TODOs in my name. Not worth splitting into
separate CLs as the changes are minor.
Bug: webrtc:9883
Change-Id: I05c54b76507a1d51b92cad080ca4e2dfe8546bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155520
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29377}
This will reduce bias caused by uncertainty in averaging window.
Bug: None
Change-Id: I5c4fe39ffe69fb4af87d86995196a54115d3e0b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144720
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29374}
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total allocatable bitrate.
Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.
It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.
Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
The header files
api/congestion_control_interface.h
api/data_channel_transport_interface.h
api/datagram_transport_interface.h
api/media_transport_config.h
api/media_transport_interface.h
have been moved into the api/transport/ and api/transport/media
subdirectories.
Bug: webrtc:8733
Change-Id: I98752c4d1306b54559bafa71712b105932c08834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153522
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29357}
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}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
Adds a field trial and configuration parameter to control whether
datagram transport may be used for data channels in a receive-only
manner. By default, if use_datagram_transport_for_data_channels is
enabled, PeerConnection will create a datagram transport and offer its
use for outgoing calls as well as accept incoming offers with compatible
datagram transport parameters.
With this change, a receive_only mode is added for datagram transport
data channels. When receive_only is set, the PeerConnection will not
create or offer datagram transports for outgoing calls, but will accept
incoming calls that offer compatible datagram transport parameters.
Bug: webrtc:9719
Change-Id: I35667bcc408ea4bbc61155898e6d2472dd262711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154463
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29327}
This is a reland of 487f9a17e4
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.
Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
Only the ISAC codec had an non-trivial implementation, for its unused
adaptive mode. This cl deletes that implementation, and the call
from NetEq, and the interface method.
Bug: webrtc:10098
Change-Id: Iaf7667e0ae867fc9d64286dff4c01a8ce0b6e2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29279}
Transport parameters are no longer retreived using this method, and no
implementations currently override it.
Bug: webrtc:9719
Change-Id: Iba0e1c7a320266f199aab6f2add36c6a22b48458
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154004
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29272}
These methods are implemented everywhere, and no longer need to provide
default implementations.
Bug: webrtc:9719
Change-Id: I2b33ace17696ec832a9936cf02a81c4973158046
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154003
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29263}
These methods are implemented everywhere, so they no longer need to
provide default implementations.
Bug: webrtc:9719
Change-Id: Idf67a78010a55f545d882793d0d6edbccfae525b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154002
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29262}
Returning absl::string_view causes problems to the Chromium/WebRTC
component build because absl::operator<< needs to be exported.
This CL switches to `const char*` which should be enough to avoid
to generate temporaries.
Bug: webrtc:9419
Change-Id: If169a6f95c7efd21ac8ce108c7f2f80a76ff2313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153842
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29250}
Keeping default implementations only for methods involved in
ongoing transitions.
Intended to catch inconsistencies between the interface and the
PeerConnectionProxy class, at compile time.
Bug: webrtc:10716
Change-Id: I4cb126c353855f7288ba09273fa6f87aaa0f32eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29224}
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}
Follow-up of https://webrtc-review.googlesource.com/c/src/+/153220,
where during code review it was suggested to move webrtc::MinPositive
out of the api/ directory.
Bug: None
Change-Id: I0c3b87a9ffd1cd205a85dddd9f44cfd95eb02206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29220}
Extend PeerConnectionE2EQualityTestFixture::VideoConfig with
min_encode_bitrate_bps and max_encode_bitrate_bps.
These are needed to be able to specify the bitrate to be used in tests.
Bug: None
Change-Id: I8af88020e9b364d924e2cecb2bdcc12bf287394d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153352
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29219}
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}
from 2x expected time to 10x.
To decrease flakiness for task queue implemntations that destroy tasks
after destruction of the task queue.
Bug: chromium:1000531
Change-Id: Ieb37ff782ead585e0aa2c84472e3993107c5c072
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152830
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29204}
Will be populated in a later cl.
Bug: webrtc:8733
Change-Id: I7e136645380d2264697c72f2d49403b3b9f0f044
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153341
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29203}
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}
Intended as a utility base class for tests, to make it easier to
delete default implementations of PeerConnectionInterface methods.
Bug: webrtc:10716
Change-Id: Ie125747ad88d209c4797cc13253aef61275ed7b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152820
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29184}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
The NetworkStateEstimator is updated on every incoming RTP packet if available.
A rtcp::RemoteEstimate packet is sent every time a rtcp::TransportFeedback packet is sent.
BUG=webrtc:10742
Change-Id: I4cd8e9d85d35faf76aeefd2e26c2a9fe1a62ca3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152161
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29143}
Multichannel signals are downmixed to mono before decimation and
delay estimation. This is useful when not all channels play
audio content. The feature can be toggled in the AEC3 configuration.
Bug: webrtc:10913
Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29126}
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}
I gave up on removing proxy_info, user_agent and tcp_options. I don't
think it's feasible to remove them without removing all the proxy code.
The assumption that you can set the proxy and user agent long after
you have created the factory is entrenched in unit tests and the code
itself. So is the ability to set tcp opts depending on protocol or
endpoint properties.
It may be easier to untangle proxy stuff from the factory later,
when it becomes a more first-class citizen and isn't passed via
the allocator.
Requires https://chromium-review.googlesource.com/c/chromium/src/+/1778870
to land first.
Bug: webrtc:7447
Change-Id: Ib496e2bb689ea415e9f8ec1dfedff13a83fa4a8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150799
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29091}
This is a reland of a66395e72f
Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
>
> This is a reland of f3a197e553
>
> Original change's description:
> > Add core multi-channel pipeline in AEC3
> > This CL adds basic the basic pipeline to support multi-channel
> > processing in AEC3.
> >
> > Apart from that, it removes the 8 kHz processing support in several
> > places of the AEC3 code.
> >
> > Bug: webrtc:10913
> > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29017}
>
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}
Bug: webrtc:10913
Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29042}
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
This is a reland of f3a197e553
Original change's description:
> Add core multi-channel pipeline in AEC3
> This CL adds basic the basic pipeline to support multi-channel
> processing in AEC3.
>
> Apart from that, it removes the 8 kHz processing support in several
> places of the AEC3 code.
>
> Bug: webrtc:10913
> Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29017}
Bug: webrtc:10913
Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29034}
This reverts commit f3a197e553.
Reason for revert: Speculative revert, as this may'be broken some build bots
Original change's description:
> Add core multi-channel pipeline in AEC3
> This CL adds basic the basic pipeline to support multi-channel
> processing in AEC3.
>
> Apart from that, it removes the 8 kHz processing support in several
> places of the AEC3 code.
>
> Bug: webrtc:10913
> Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29017}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I877d2993b9ccf024bd1d57bca1513c3e24d0bed3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29022}
If somehow buffer is shared between other locations, reallocating it may
lead to use-after-free error.
Bug: none
Change-Id: I01a0b722cfe6ee0e18546248f1dfb7b8ac3b7217
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150884
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29021}
This CL adds basic the basic pipeline to support multi-channel
processing in AEC3.
Apart from that, it removes the 8 kHz processing support in several
places of the AEC3 code.
Bug: webrtc:10913
Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29017}
Also annotate a few of the remaining uses, to guide further splits of
that large build target.
Bug: webrtc:8733
Change-Id: I16ac33ab48e6d39a1a8dbc2a3fc671d8db6dbfe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150789
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29001}
The new target does not depend on libjingle_peerconnection_api, and to
do this, the named "audio" and "video" string literals had to be moved from
media_stream_interface.cc to media_types.cc.
In this cl, the dependency on libjingle_peerconnection_api can be
dropped from a few targets.
No-Presubmit: True
Bug: webrtc:8733
Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28998}
This patch adds accounting and reporting needed for
newly added RTCTransportStats.selectedCandidatePairChanges,
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairchanges
a) P2PTransportChannel counts everytime selected_connection_
is modified and reports this counter in the GetStats()-call.
b) RTCStatsCollector puts the counter into the standardized
stats object.
Bug: webrtc:10900
Change-Id: Ibaeca18706b8edcbcb44b0c6f2754854bcb545ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149830
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28987}
This patch adds a new (optional) attribute to TURN_ALLOCATE_REQUEST,
TURN_LOGGING_ID (0xFF05).
The attribute is put into the comprehension-optional range
so that a TURN server should ignore it if it doesn't know if.
https://tools.ietf.org/html/rfc5389#section-18.2
The intended usage of this attribute is to correlate client and
backend logs.
Bug: webrtc:10897
Change-Id: I51fdbe15f9025e817cd91ee8e2c3355133212daa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149829
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28966}
This CL removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.
Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
Deprecated the field BitrateAllocationUpdate::link_capacity since it is only
used by the Opus codec in order to smooth the target bitrate, which is
equivalent to the stable_target_bitrate field.
The unused field trial WebRTC-Bwe-StableBandwidthEstimate is also removed.
Bug: webrtc:10126
Change-Id: Ic4a8a9ca4202136d011b91dc23c3a27cfd00d975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149839
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28941}
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels. There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.
PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.
Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks. This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state. This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.
For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.
Datagram transport use is negotiated. As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer. If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport. If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.
If DTLS is not enabled, there is no viable fallback. In this case, the data
channels are negotiated as media transport data channels. If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.
Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
This is a reland of 2b9fa09fa3.
It got reverted because I forgot to whitelist the new metrics in chromium,
which has now been done:
https://chromium-review.googlesource.com/c/chromium/src/+/1760209
Relanding requires no changes to the CL.
Original change's description:
> [GetStats] Expose video codec implementation in standardized metrics.
>
> Spec issue: https://github.com/w3c/webrtc-stats/issues/445
> Spec PR: https://github.com/w3c/webrtc-stats/pull/473
>
> Now that the spec's RTCCodecStats.implementation has moved to
> RTCOutboundRtpStreamStats.encoderImplementation and
> RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
> using the same string that the legacy getStats() API used.
>
> Bug: webrtc:10890
> Change-Id: Ic43ce44735453626791959df3061ee253356015a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28877}
TBR=ilnik@webrtc.org
Bug: webrtc:10890
Change-Id: Ib874b608856c2795b1ca08f6af43c61dd859ea21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149800
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28887}
This reverts commit 2b9fa09fa3.
Reason for revert: speculative revert since it seems to break Chrome FYI bots. See https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4206
Original change's description:
> [GetStats] Expose video codec implementation in standardized metrics.
>
> Spec issue: https://github.com/w3c/webrtc-stats/issues/445
> Spec PR: https://github.com/w3c/webrtc-stats/pull/473
>
> Now that the spec's RTCCodecStats.implementation has moved to
> RTCOutboundRtpStreamStats.encoderImplementation and
> RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
> using the same string that the legacy getStats() API used.
>
> Bug: webrtc:10890
> Change-Id: Ic43ce44735453626791959df3061ee253356015a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28877}
TBR=ilnik@webrtc.org,hbos@webrtc.org
Change-Id: Ia0b7f9806564cf28881c50d6371b8141a22e3431
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10890
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149175
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28879}
Spec issue: https://github.com/w3c/webrtc-stats/issues/445
Spec PR: https://github.com/w3c/webrtc-stats/pull/473
Now that the spec's RTCCodecStats.implementation has moved to
RTCOutboundRtpStreamStats.encoderImplementation and
RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
using the same string that the legacy getStats() API used.
Bug: webrtc:10890
Change-Id: Ic43ce44735453626791959df3061ee253356015a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28877}
This CL will make AudioDecoderIsacT symmetrical to AudioEncoderIsacT.
Bug: webrtc:10826
Change-Id: I78d1cf7bc2245bf4a282aabd81c8ece6ca23f285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28847}
DataChannelTransportInterface takes the OpenChannel, SendData,
CloseChannel, and SetDataSink methods. MediaTransportInterface inherits
from DataChannelTransportInterface.
DatagramTransportInterface, the newer alternative to
MediaTransportInterface, also inherits from
DataChannelTransportInterface.
This will allow further refactors to enable the use of media-transport
style data channels alongside the datagram transport.
Bug: webrtc:9719
Change-Id: I2dd873785ea52d38055b62545c17e9e17c4e70c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147840
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28846}
This CL adds the following options:
pass an input AEC dump as a string (currently, the tool can only accept a path to an AEC dump file)
write the processed capture samples to a given vector
Bug: webrtc:10808
Change-Id: I02863c97ec3cd8c03ade2ea8521836f2e7417050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145208
Commit-Queue: Sonia-Florina Horchidan <soniahorchidan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28826}
This CL removes a long unused fallback behavior for the reverb
computation.
Bug: webrtc:8671
Change-Id: I4b57795a9bb33769237858f40392881ee235653e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148520
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28802}
This is a reland of 0a88ea050c.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.
Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
This CL removes code related to the usage of the delay agnostic and
extended filter modes in AEC2.
Bug: webrtc:8671
Change-Id: I1a2c7a9eba54b03f5a015df3adb617785f52a939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133912
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28789}
In order to be able to detect and measure context around candidate pair changes.
Bug: webrtc:10419
Change-Id: Iab0d7e7c80d925d1aa44617fc35975fdc6bbc6b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147340
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28779}
This reverts commit 7f1c58938d.
Reason for revert: this has been temporarily postponed.
Original change's description:
> Adding new top-level directory crypto/
>
> Adding the crypto root directory to WebRTC. The goal with this change is to
> centralize the management of crypto code into a single location.
>
> Currently we have cryptography code scattered across pc/ and rtc_base/
> which makes it difficult audit and maintain.
>
> By having a crypto/ directory we gain:
> 1. A clear first point of contact for auditing the cryptography in WebRTC.
> 2. Fine grain ownership to cryptography maintainers, we can include BoringSSL
> maintainers in this directory.
> 3. It improves maintanability of crypto code as we have improved modularization.
> It will not be deeply nested in all different parts of WebRTC.
> 4. Improved testability. We can cleanly build crypto libraries which plug into
> pc/ which we can more easily mock.
> 5. Enforce stricter rules. For example we may want to enforce ZeroOnFreeBuffer
> for all sensitive material. This is easier to enforce in a single directory.
>
> Bug: webrtc:9600
> Change-Id: I8e76332c7dcdac0a45a470ba2e930196e1ccf395
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125142
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27028}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,benwright@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9600
Change-Id: I3c99e733d53d76071179f0ff9ffdec965d20829d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147871
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28750}
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}
It's planned to be deprecated so it should not be required.
Bug: webrtc:9883
Change-Id: I7daa922786d3cbf6bca38e205f4f57773f3f8448
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147275
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28715}
This reverts commit fab3460a82.
Reason for revert: fix downstream instead
Original change's description:
> Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
>
> This reverts commit 9973933d2e.
>
> Reason for revert: breaking downstream projects and not reviewed by direct owners
>
> Original change's description:
> > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> >
> > This reverts commit 24192c267a.
> >
> > Reason for revert: Analyzed the performance regression in more detail.
> >
> > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f6 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac.
> >
> > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall.
> >
> > Original change's description:
> > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> > >
> > > This reverts commit 3e8ef940fe.
> > >
> > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
> > >
> > > Original change's description:
> > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
> > > >
> > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
> > > >
> > > > Bug: webrtc:10668
> > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > > > Commit-Queue: Chen Xing <chxg@google.com>
> > > > Cr-Commit-Position: refs/heads/master@{#28434}
> > >
> > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
> > >
> > > Bug: webrtc:10668, chromium:982260
> > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
> > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28561}
> >
> > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: webrtc:10668, chromium:982260
> > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28664}
>
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
>
> Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10668, chromium:982260
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28671}
TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10668, chromium:982260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28672}
This reverts commit 9973933d2e.
Reason for revert: breaking downstream projects and not reviewed by direct owners
Original change's description:
> Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
>
> This reverts commit 24192c267a.
>
> Reason for revert: Analyzed the performance regression in more detail.
>
> Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f6 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac.
>
> There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall.
>
> Original change's description:
> > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> >
> > This reverts commit 3e8ef940fe.
> >
> > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
> >
> > Original change's description:
> > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
> > >
> > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
> > >
> > > Bug: webrtc:10668
> > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > > Commit-Queue: Chen Xing <chxg@google.com>
> > > Cr-Commit-Position: refs/heads/master@{#28434}
> >
> > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
> >
> > Bug: webrtc:10668, chromium:982260
> > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28561}
>
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10668, chromium:982260
> Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Chen Xing <chxg@google.com>
> Cr-Commit-Position: refs/heads/master@{#28664}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10668, chromium:982260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28671}
This reverts commit 24192c267a.
Reason for revert: Analyzed the performance regression in more detail.
Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f6 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac.
There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall.
Original change's description:
> Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
>
> This reverts commit 3e8ef940fe.
>
> Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
>
> Original change's description:
> > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
> >
> > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
> >
> > Bug: webrtc:10668
> > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28434}
>
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
>
> Bug: webrtc:10668, chromium:982260
> Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28561}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10668, chromium:982260
Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28664}
SVC support is limited:
During SVC testing there is no SFU, so framework will try to emulate SFU
behavior in regular p2p call. Because of it there are such limitations:
* if |target_spatial_index| is not equal to the highest spatial layer
then no packet/frame drops are allowed.
If there will be any drops, that will affect requested layer, then
WebRTC SVC implementation will continue decoding only the highest
available layer and won't restore lower layers, so analyzer won't
receive required data which will cause wrong results or test failures.
Bug: webrtc:10138
Change-Id: I079566260ca9f1815935bce365d1bca10766663a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144882
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28612}
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
This is part of "Perfect Negotiation" (https://crbug.com/980872).
Spec PR here (merged): https://github.com/w3c/webrtc-pc/pull/2169
Spec: https://w3c.github.io/webrtc-pc/#dfn-localufragstoreplace
The restartIce() makes the next createOffer() generate new ICE
credentials, as if {iceRestart:true} was passed in as options. It also
causes negotiationneeded. This is better than manually restarting ICE
because it survives rollbacks (when that is implemented) and
restartIce() can be called regardless of current signalingState.
Bug: chromium:980881
Change-Id: I8e70bec31ce9d4d6a303bd35e91b2dcc28fcad60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144941
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28596}
This reverts commit 3e8ef940fe.
Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
Original change's description:
> Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
>
> This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
>
> Bug: webrtc:10668
> Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Chen Xing <chxg@google.com>
> Cr-Commit-Position: refs/heads/master@{#28434}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
Bug: webrtc:10668, chromium:982260
Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28561}
The GetImplementations function is similar to the GetSupportedFormats function, but instead of providing one SdpVideoFormat per codec it provides one per codec implementation. These SdpVideoFormats can then be tagged so that a certain implementation can be instantiated when CreateVideoEncoder is called.
Bug: webrtc:10795
Change-Id: I79f2380aa03d75d5f9f36138625abf3543c2339d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145215
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28553}
This change removes RTCP Feedback loop if we are using datagram transport by removing transport sequence numbers from RTP packets and recreating RTCP Feedback from Datagram ACKs and Timestamps.
- For outgoing RTP packets, remove transport sequence number and store it with datagram_id. Note that removing transport sequence numbers does not only save 4-8 bytes per packet, but also prevents generation of feedback packets on the receiver side.
- When datagram ACKs, we re-created RTCP feedback with timestamp.
- Replacing previous assumption that datagram_id was the same as packet_id by storing incremental counter of datagram ids (I noticed some packets come without packet_id, which is a bit strange, but easy to support and it's also good not to rely on packet_ids being unique across multiple ssrcs).
Bug: webrtc:9719
Change-Id: Iecfe938ecea1a74e7c9e1484f0e985d72643d4a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145269
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28542}
Users of webrtc generally should be able to choose own task queue implementation.
Poison avoids accidental dependency of a low level component on the default implementation
Android and ios apis are still de-facto forced to use the default implementation.
Bug: webrtc:10284
Change-Id: I67ecf2317f43ee32b0c9e8a6e69f1e0987cf1914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144786
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28524}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
Ntp timestamps are used for end-to-end delay measurements and can never
go away. The naming and number of timestamp fields in VideoFrame could
change in the future, but capture time in local clock will always be
there on the receive side.
Bug: none
Change-Id: I358689cd8a44b1da8503136b3dd898b936f2d693
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144542
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28498}
There is no public API to create NetworkBehaviorInterface from
BuiltInNetworkBehaviorConfig, so this CL will add direct method, that will
allow downstream projects to use BuiltInNetworkBehaviorConfig for network
emulation.
Bug: webrtc:10138
Change-Id: Iaec3ea17c12bd06b1c0ff3e5bc2b32cc1c4f62f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144628
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28494}
Because of webrtc:10801, we don't actually support 4 simulcast layers but 3.
Until this is fixed, we limit the value to what we can currently handle.
Bug: webrtc:8785
Change-Id: I513b7c8d4c889fa0d80c91adc1c4f874acb86fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144625
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28487}
Two new classes are added to WebRTC from Chrome: ChannelMixer and
ChannelMixingMatrix but they are not yet utilized in the audio path for
WebRTC.
The idea is to utilize these new classes when adding support for multi-
channel encoding/decoding in WebRTC/Chrome.
Adds support for a new enumerator call webrtc::ChannelLayout and some
helper methods which maps between channel layout and number of channels.
These parts are also copied from Chrome.
Minor (cosmetic) changes are also done on the AudioFrame to prepare
for upcoming work.
Bug: webrtc:10783
Change-Id: I6cd7a13a3bc1c8bbfa19bc974c7a011d22d19197
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141674
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28482}
Implements RTCAudioSourceStats members:
- audioLevel
- totalAudioEnergy
- totalSamplesDuration
In this CL description these are collectively referred to as the audio
levels.
The audio levels are removed from sending "track" stats (in Chrome,
these are now reported as undefined instead of 0).
Background:
For sending tracks, audio levels were always reported as 0 in Chrome
(https://crbug.com/736403), while audio levels were correctly reported
for receiving tracks. This problem affected the standard getStats() but
not the legacy getStats(), blocking some people from migrating. This
was likely not a problem in native third_party/webrtc code because the
delivery of audio frames from device to send-stream uses a different
code path outside of chromium.
A recent PR (https://github.com/w3c/webrtc-stats/pull/451) moved the
send-side audio levels to the RTCAudioSourceStats, while keeping the
receive-side audio levels on the "track" stats. This allows an
implementation to report the audio levels even if samples are not sent
onto the network (such as if an ICE connection has not been established
yet), reflecting some of the current implementation.
Changes:
1. Audio levels are added to RTCAudioSourceStats. Send-side audio
"track" stats are left undefined. Receive-side audio "track" stats
are not changed in this CL and continue to work.
2. Audio level computation is moved from the AudioState and
AudioTransportImpl to the AudioSendStream. This is because a) the
AudioTransportImpl::RecordedDataIsAvailable() code path is not
exercised in chromium, and b) audio levels should, per-spec, not be
calculated on a per-call basis, for which the AudioState is defined.
3. The audio level computation is now performed in
AudioSendStream::SendAudioData(), a code path used by both native
and chromium code.
4. Comments are added to document behavior of existing code, such as
AudioLevel and AudioSendStream::SendAudioData().
Note:
In this CL, just like before this CL, audio level is only calculated
after an AudioSendStream has been created. This means that before an
O/A negotiation, audio levels are unavailable.
According to spec, if we have an audio source, we should have audio
levels. An immediate solution to this would have been to calculate the
audio level at pc/rtp_sender.cc. The problem is that the
LocalAudioSinkAdapter::OnData() code path, while exercised in chromium,
is not exercised in native code. The issue of calculating audio levels
on a per-source bases rather than on a per-send stream basis is left to
https://crbug.com/webrtc/10771, an existing "media-source" bug.
This CL can be verified manually in Chrome at:
https://codepen.io/anon/pen/vqRGyq
Bug: chromium:736403, webrtc:10771
Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28480}
This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at:
http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
We are still missing the code to:
- Negotiate the header extension.
- Collect capture time for audio and video and have the info sent with the header extension.
- Receive the header extension and use its info.
Bug: webrtc:10739
Change-Id: I75af492e994367f45a5bdc110af199900327b126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28468}
In this CL:
- Added WEBRTC_VIDEO_CODEC_ENCODER_FAILURE return code that can
be returned by the encoder wrapper in case of a broken encoder.
- Added EncoderFailureCallback interface that can be called
to request encoder fallback to be performed. Implemented by
WebRtcVideoChannel and called from the VideoStreamEncoder.
- Updated SelectSendVideoCodec to select all compatible codecs instead
of just one.
Bug: webrtc:10795
Change-Id: I87a83fd02e48c40493c930471c06c3d0941031ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140888
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28462}
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}
First, the existing configuration parameter (use_datagram_transport) is
now optional.
The new field trial has two flag values:
1. Whether to enable the datagram transport (enabled)
2. Whether to use the datagram transport by default (default_value)
The first is a kill-switch. It disables the datagram transport, even
for applications which inject a datagram transport factory and specify
use_datagram_transport = true. This allows applications which hard-code
a datagram transport to switch it off via field trials.
This flag defaults to true, to avoid breaking downstream projects which
already inject and configure a datagram transport. It may be changed to
false after updating downstream to set this field trial flag to true
when required.
The second provides a default value to be used in case the
aforementioned use_datagram_transport parameter is unset. Applications
which explicitly set use_datagram_transport will use that value.
Applications which do not explicitly specify whether or not to use the
datagram transport will use it (or not) according to the default_value
flag.
One goal of this flag is to simplify rollout in applications which
already set field trials based on configuration, but require code
changes for new RTCConfiguration parameters. A second goal is to
provide platforms with a knob to control whether datagram transport is
"opt-in" or "opt-out".
This flag defaults to false, to prevent downstream projects from
unintentionally enabling the datagram tranpsort.
Bug: webrtc:9719
Change-Id: I521a5fa61c992e76e5081118678a1812a261d672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28435}
This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28434}
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.
This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
via this API.
TBR=kwiberg@webrtc.org
Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
This is part of a series of CLs with the goal to make the network
state estimate fields more semantically useful.
Bug: webrtc:10742
Change-Id: I3185dc8c4574710a4dce44a4ffd6cefe0fb4e8cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144036
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28406}
Bug: None
Change-Id: I0c82815b21b1eb0be3e12cf6ad52bf6082dfea7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143798
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28392}
When provided, these thresholds will be used instead of WebRTC default
limits specified in DropDueToSize() and GetMaxDefaultVideoBitrateKbps().
Bug: none
Change-Id: Ida45ea832041963b8b8475d69114b5c60a172fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142170
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28390}
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28385}
This is a reland of fa79081dca
It crashed due to inability to handle small timestamps in probe
estimator. This was fixed by moving history window check to avoid
subtracting from the timestamp.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
Bug: webrtc:9510
Change-Id: I3bf91222068e4fbb6aa159bfeb7a73e00bb6a0d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143165
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28347}
This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}
This reverts commit fa79081dca.
Reason for revert: Breaks downstream project.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
TBR=terelius@webrtc.org,srte@webrtc.org
Change-Id: I562365fc5d1da68326d603338ccc6371114d7e12
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9510
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143164
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28331}
As the send time congestion controller now has been removed,
we don't need the RTP related constructs anymore.
Bug: webrtc:9510
Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28330}
Users that would like custom TaskQueueFactory (e.g. chromium) should use
CreateModularPeerConnectionFactory directly
Bug: webrtc:10284
Change-Id: I7ba55a0f21560d4beb71a7f93f6fa70b0fe74931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142234
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28324}
This change is breaks the dependency between "api:rtp_headers" and "api/video:video_frame".
Bug: webrtc:10668
Change-Id: Ib827de2c0e33809ab1d58338037563aa2fa249bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140949
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28322}
This is part of a series of CLs removing RTP dependencies from GoogCC
implementation.
Bug: webrtc:10749
Change-Id: I4d8ec92643f1a0d4b5fdac2fbcb067766afc6548
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142222
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28300}
