By enabling the field trial, REMB caps the output target bitrate, but
does not change any internal BWE state variables.
Bug: webrtc:12306
Change-Id: I43e9ac1d1b7dff292d7aa5800c01d874bc91aaff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197809
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32867}
The previous tests ran in real-time making them flaky, so they were
disabled on a number of platforms.
This CL ports the tests 1:1 (sort of) to use the scenario test
framework which runs with simulated time and much less risk of
flakiness.
Bug: webrtc:10155
Change-Id: I6281f57d73883c8aaa91964e9cfa58d9b47779fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32333}
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}
https://webrtc-review.googlesource.com/c/src/+/172847
------------ original description --------------
Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
This reverts commit 24eed2735b.
Reason for revert: Speculative revert: breaks downstream project
Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
>
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
>
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
>
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
>
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
>
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
>
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}
TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
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}
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.
Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
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}
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}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
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}
This API is going away, we'll use the WebRTC-Audio-Allocation field
trial flag to set this value in the future.
Bug: webrtc:10556
Change-Id: I2c4c1948a33f909fac069dd038cea36a793e4745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145405
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28608}
Simplifying the code to better fit with how it is used.
Bug: webrtc:9883
Change-Id: I2bd52f26b829413e516dee4f551cf36574275019
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136681
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27994}
Removing functionality to choose congestion controller implementation,
using injection instead. Also cleaning up some related functionality
that's no longer needed, such as the injection of event logs into the
factory.
Bug: webrtc:9883
Change-Id: Ia528005625430ae31a15bc88881e2d4ac6ad1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133890
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27768}
Also adding sync group for video streams.
Bug: webrtc:10365
Change-Id: I9ef92de756f06bbbcd7b67524bbf51fe1365fa85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27390}
As part of this change, a task queue is used to handle packet
processing in real time mode. This requires that we also do
most call and media stream related operation on the same task
queue to satisfy thread checkers.
Bug: webrtc:10365
Change-Id: Icdd9d56e4ca14f2c944dc655c91e29392e3765f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127544
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27379}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
This makes it possible to save log outputs from scenario tests to
either files or memory.
Bug: webrtc:9510
Change-Id: I883bd8240ab712d31d54118adf979041bd83481a
Reviewed-on: https://webrtc-review.googlesource.com/c/116321
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26284}
It is a step in the big refactoring to introduce new network emulation
layer for peer connection level e2e test, which will be based on system
sockets level injection.
Bug: webrtc:10138
Change-Id: Ie3854d22aa3eec289617bc432026ea670646556a
Reviewed-on: https://webrtc-review.googlesource.com/c/115943
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26137}
The return value is not used. This change prepares for future
refactoring by removing the requirement that TryDeliverPacket must be
synchronous. Also renaming to DeliverPacket as we no longer need to
indicate the meaning of the return value.
Bug: webrtc:9510
Change-Id: I78536434b198fa7bf4df88b10d6add23684767f1
Reviewed-on: https://webrtc-review.googlesource.com/c/115181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26066}
This makes it possible to test custom network controllers without
requiring update to test framework. Also updating BBR performance
test to use this feature.
Bug: webrtc:9510
Change-Id: I0446de0403fe9d1f6dc3710c1d114887a6c359c5
Reviewed-on: https://webrtc-review.googlesource.com/c/114640
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26046}
This prevents printing warning messages when the extensions aren't
found. The real parsing is done deeper in the stack and is unaffected.
Bug: webrtc:9510
Change-Id: Idf09f0e69c223bd4217be7044d21d1d0bbbdab92
Reviewed-on: https://webrtc-review.googlesource.com/c/110615
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25612}
This CL makes it possible to change transport routes while running
a scenario based test.
To make this possible in a consistent manner, the scenario test
framework is modified to only allow shared transport for all streams
between two CallClients. This is what typically is done in practice and
it is quite complex to even reason about the implications of using
mixed transports for a single call.
Bug: webrtc:9718
Change-Id: Ib836928feed98aa2bbbe0295e158157a6518348b
Reviewed-on: https://webrtc-review.googlesource.com/c/107200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25287}
