Replaces both BitrateConstraintsMask and
PeerConnectionInterface::BitrateParameters. The latter is kept
temporarily for backwards compatibility.
Bug: None
Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f
Reviewed-on: https://webrtc-review.googlesource.com/74020
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23148}
This CL extracts the part of FakeNetworkPipe responsible for simulating
network behavior into the SimulatedNetwork class, which implements the
new FakeNetworkInterface.
This prepares for an upcoming CL where the network simulation can
be injected in FakeNetworkPipe, allowing custom simulation models to be
used.
Bug: None
Change-Id: I9b5fa0dd9ff1fd8ccd5a7ce2d9ea3a5b11c5215e
Reviewed-on: https://webrtc-review.googlesource.com/64405
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23146}
Moving ownership of worker task queue in Call to
RtpTransportControllerSend. This CL also ensures that the task queue
is not destroyed until the process thread running
SendSideCongestionController is stopped.
The worker queue should be owned by RtpTransportControllerSend since
it is mainly used for rtp and transport related tasks such as bitrate
allocation and signaling network state.
Bug: webrtc:9232
Change-Id: I211edf1a3b9f9b2572875d5584cb788cb2449ef9
Reviewed-on: https://webrtc-review.googlesource.com/63023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23119}
To make this work, also make the private method
GetTransmissionMaxBitrateMultiplier static.
Bug: None
Change-Id: I1777602b89d404ecbc9956acf9a9c7f852d69911
Reviewed-on: https://webrtc-review.googlesource.com/72440
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23023}
This CL changes the usages of milliseconds as a unit in FakeNetworkPipe
to microseconds. This matches the time unit of the PacketTime struct and
increases the precision of the simulation. The time resolution in
FakeNetworkPipe::Config is kept unchanged to keep the values more human
readable.
This CL prepares refactoring in upcoming CLs.
Bug: webrtc:9054
Change-Id: I103f7a0afa41381f676ea07fcc8c083532e61f1d
Reviewed-on: https://webrtc-review.googlesource.com/64140
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23018}
This CL moves the responsibility for demuxing from FakeNetworkPipe
to DirectTransport. This makes the interface for FakeNetworkPipe more
consistent. It exposes fewer different interfaces for different usages.
It also means that any time degradations applied to the packets due in
FakeNetworkPipe in tests will now be propagated to Call in a more
realistic manner. Previously the time was set to uninitialized which
meant that Call filled in values based on the system clock.
Bug: webrtc:9054
Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0
Reviewed-on: https://webrtc-review.googlesource.com/64260
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23017}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
This is used to allow mocking in tests in later CLs.
Bug: None
Change-Id: Id610471efb4a86c903530585dd4ee2fa1d1ea5bc
Reviewed-on: https://webrtc-review.googlesource.com/70880
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22941}
This makes it easier to refactor the interface in upcoming CLs.
Bug: None
Change-Id: I67d0216e24f087294e95ac96f7278f302bf69832
Reviewed-on: https://webrtc-review.googlesource.com/71041
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22933}
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This ensures that packet time effects are simulated properly when using
a demuxer with FakeNetworkPipe. Previously this was only done using a
receiver.
This prepares for a CL with a test depending on this behavior.
Bug: webrtc:9054
Change-Id: I031acc9e18adc2891d3e396352dcd2614211909c
Reviewed-on: https://webrtc-review.googlesource.com/67342
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22918}
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.
Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
In preparation for moving creation of encoders to VideoStreamEncoder;
then this field is needed for calling
VideoEncoderFactory::CreateVideoEncoder.
Bug: webrtc:8830
Change-Id: Ie107a93596e22a5e7b9a9147bd85a93cf84b15a3
Reviewed-on: https://webrtc-review.googlesource.com/70221
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22894}
The SendSideCongestionController version is toggled by an experiment in
RtpTransportControllerSend. This CL adds a log statement of which
version is used, to make debugging easier.
Bug: webrtc:8415
Change-Id: I6201cf5f03e097cc07c6ae120dcff075c046c414
Reviewed-on: https://webrtc-review.googlesource.com/69808
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22858}
Added functionality on the FakeNetworkPipe to introduce arbitrary
clock offsets. This offset is added to the reported receive time of
all packets. This prepares for a later CL using this to test correction
of receive time stamps.
Bug: webrtc:9054
Change-Id: I811b3aa8359bc917f59443088d8a418368242db9
Reviewed-on: https://webrtc-review.googlesource.com/64726
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22763}
This ensures that the callback will be called if total max bit rate
changes even if min bitrate or padding bitrate has not changed.
Bug: None
Change-Id: I616e95b1f9f5a30733f1d0acb86e18c93001d3db
Reviewed-on: https://webrtc-review.googlesource.com/63642
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22734}
The MID header extension is handled by the RtpTransport
which lives above Call and takes care of demuxing to SSRC.
Bug: webrtc:4050
Change-Id: I27135e296ae9c7b15e926ba17547c26c75684ad6
Reviewed-on: https://webrtc-review.googlesource.com/65025
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22682}
This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
Reason for revert: Regression in ramp up perf tests.
Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
Reduce synchronization in the class significantly and not hold a lock
while calling out to external implementations.
* Rewrite tests to use a real ProcessThread.
* Update some code to use C++ 11 constructs & library features.
Bug: webrtc:9064
Change-Id: I240a819efb6ef8197da3f2edf7acf068d2a27e8b
Reviewed-on: https://webrtc-review.googlesource.com/64521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22649}
In preparation for also moving the responsibility for encoder creation.
Bug: webrtc:8830
Change-Id: Ic3d2039a86cd31c1b4157f5df4e97b607c81f1d7
Reviewed-on: https://webrtc-review.googlesource.com/55264
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22630}
Modifies RtpTransportControllerSend to store a raw pointer to send side
congestion controller(SSCC). This avoids a race between destruction of
the send_side_cc_ unique pointer and calling AvailableBandwidth on
the SSCC instance from the OnNetworkChanged callback.
Bug: None
Change-Id: I11f414d7db48ab0b29a049b9488b073c1551113d
Reviewed-on: https://webrtc-review.googlesource.com/64640
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22604}
This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
Bug: webrtc:8830
Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
Reviewed-on: https://webrtc-review.googlesource.com/63721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22595}
Replacing "/" with "," as separator. "/" is not allowed as a character
in field trial strings.
Bug: webrtc:9054
Change-Id: I505da51096e3b8b357febfb9642471e3327e98e4
Reviewed-on: https://webrtc-review.googlesource.com/64444
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22591}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
NOPRESUBMIT=true
Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425
Reviewed-on: https://webrtc-review.googlesource.com/21328
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22587}
This CL adds reporting of per packet feedback availability from Call
via RtpTransportControllerSend to SendSideCongestionController.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I20b3dbb4a027c46476bc2d2bc875374bff05609a
Reviewed-on: https://webrtc-review.googlesource.com/63220
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22566}
Another step of the transition needed to reland cl
https://webrtc-review.googlesource.com/62062, and move payload_name
and payload_type out of VideoSendStream::Config::EncoderSettings.
If the new fields are set, values of the old fields are ignored.
Bug: webrtc:8830
Change-Id: I1f0cd56fd6b13b05608b284afc92523707887e25
Reviewed-on: https://webrtc-review.googlesource.com/64101
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22562}
First step of the transition needed to reland cl
https://webrtc-review.googlesource.com/62062, and move payload_name
and payload_type out of VideoSendStream::Config::EncoderSettings.
If the new field is set to something different than kVideoCodecUnkown,
payload_name from EncoderSettings is ignored.
Bug: webrtc:8830
Change-Id: I515a91f8291cda79017332102cc6a10736d8a648
Reviewed-on: https://webrtc-review.googlesource.com/64001
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22555}
The receive time calculator combines the packet time stamps received
from the socket interface with the system clock used in WebRTC. This
means that the packet timestamps are set in the WebRTC clock timebase
and that large jumps in the time stamps from the socket will not affect
the reported receive time stamps.
Bug: None
Change-Id: I293925c41919829524a115bb9377027bf0a797fb
Reviewed-on: https://webrtc-review.googlesource.com/61862
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22540}
This reverts commit bc900cb1d1810fcf678fe41cf1e3966daa39c88c.
Reason for revert: Broke downstream projects.
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
This CL adds tracking of packet feedback availability in the
BitrateAllocator class.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: Ie46d7040494ba116928d791a8e4c5dae6992cafc
Reviewed-on: https://webrtc-review.googlesource.com/63202
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22534}
This CL adds a boolean indicating availability of per packet feedback
to the OnAllocationLimitsChanged callback on the
BitrateAllocator::LimitObserver interface.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I5bd6e5796733da312556f2f681ff06d49ea2becc
Reviewed-on: https://webrtc-review.googlesource.com/63201
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22533}
In VideoSendStream::Config, move payload_name and payload_type from
EncoderSettings to Rtp.
EncoderSettings now contains configuration for VideoStreamEncoder only,
and should perhaps be renamed in a follow up cl. It's no longer
passed as an argument to VideoCodecInitializer::SetupCodec.
The latter then needs a different way to know the codec type,
which is provided by a new codec_type member in VideoEncoderConfig.
Bug: webrtc:8830
Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
Reviewed-on: https://webrtc-review.googlesource.com/62062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22532}
Give internal test tools access to FakeNetworkPipe data members
by adding a set of access methods.
Also deleted copy assignment operator for NetworkPacket.
Bug: None
Change-Id: I451a21e0cc6ec82ea830cf197c7a4cef0789623c
Reviewed-on: https://webrtc-review.googlesource.com/63301
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22515}
Removing functions that has been removed from
RtpTransportControllerSendInterface from
MockRtpTransportControllerSend.
Deleted functions: GetPacerModule, GetModule,
SetTransportOverhead and AvailableBandwidth.
Bug: webrtc:8415
Change-Id: I24d460bd18d57966e3b333ce0c234c3e3dc19a9a
Reviewed-on: https://webrtc-review.googlesource.com/62762
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22489}
The task queue based SendSideCongestionController (SSCC) was accessing
a unique pointer to the task queue from the task queue itself. This
triggered a tsan check failure when resetting the same unique pointer.
Also move declaration of SSCC member in RtpTransportControllerSend last,
to ensure that it, and its TaskQueue, are destroyed before other members.
Bug: webrtc:8415
Change-Id: I75c93f41deab637f7e4766ac4b61713c86f866e9
Reviewed-on: https://webrtc-review.googlesource.com/62143
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22478}
