It didn't have proper time to stabilize, making it sensitive to small
changes. This CL increases the stabilization period from 20 to 30s.
Also fixing some minor test suite bug found during investigation.
Bug: webrtc:9718
Change-Id: If56dba5383251ad3d3efe304eebcd880522afabe
Reviewed-on: https://webrtc-review.googlesource.com/c/119943
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26408}
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.
Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
By enabling this trial, we can also remove reporting of packet
feedback status from send streams that was used before.
Bug: webrtc:9718
Change-Id: I3e7c4656b0ac6592a834617e044f23a072454181
Reviewed-on: https://webrtc-review.googlesource.com/c/118281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26363}
This cl is part of work to move several experiments into a joint
experiment group. Most of them vill be ralted to video, hence the name.
Bug: webrtc:10223
Change-Id: I8767c43abb6aa910ab51710eeb908e0f9df1e296
Reviewed-on: https://webrtc-review.googlesource.com/c/118361
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26346}
Fix has 2 parts:
1. Fix for the LossBasedControl being at much lower levels than
DelayBased in StartUpPhase.
2. Explicitly fix state machine problem leading to toggling between
the two estimates.
Bug: webrtc:10222
Change-Id: Ieaaaec6c9233da61a86b69d936c4979c79645686
Reviewed-on: https://webrtc-review.googlesource.com/c/118280
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26327}
The WebRtcKeyValueConfig interface allows providing custom key value
configurations that changes per instance of GoogCcNetworkController.
Bug: webrtc:10009
Change-Id: I520fff030d1c3c755455ec8f67896fe8a6b4d970
Reviewed-on: https://webrtc-review.googlesource.com/c/116989
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26312}
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}
That means it does not have to be set on every update of StreamsConfig.
BUG=webrtc:9586
Change-Id: I6a348160e209042857c4475323466e2aa92adef8
Reviewed-on: https://webrtc-review.googlesource.com/c/116690
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26184}
While this class is deprecated, it's needed as a stop-gap solution.
Other methods to configure the max probe rate all effect the current
estimate and/or trigger new probes to be sent, and we need a way to
configure the max without affecting other behavior.
Bug: webrtc:10070
Change-Id: I2b0ba2fef42d0bab6e5ea7f7c921681557802b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/114880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26162}
Fix a bug introduced in (https://webrtc-review.googlesource.com/c/src/+/105102) that causes cwnd pushback only active when there is network condition changes.
Bug: None
Change-Id: I8164d5663304ce2e445db09205f706011ff7d784
Reviewed-on: https://webrtc-review.googlesource.com/c/115945
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26122}
The new controller behaves mostly like before, but increases the target
rate on timer update rather than when feedback is received. This makes
the behavior easier to predict. It also uses a duration parameter to
track the increase, removing the meed for the minimum rate increase
constants that exists in the previous solution.
Bug: webrtc:9718
Change-Id: Iae31a9ba2d6474a8236f8eb72f86ff434f1d1fc6
Reviewed-on: https://webrtc-review.googlesource.com/c/114681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26088}
If we're in ALR, the acked rate is going to be significantly lower than
the current estimate for the link capacity. If we need to back off in
this situation (usually caused by latency spikes), this CL makes us back
off relative to current estimate if. We then immediately send a new
probe just in case the network did actually change.
All of this is behind experiment flags for now.
Bug: webrtc:10144
Change-Id: I062a259c36417eea2211d44592ef7fc979aa22b7
Reviewed-on: https://webrtc-review.googlesource.com/c/113880
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26045}
The removed coded causes problems if the same RTCP packet is forwarded
to the congestion controller multiple times.
Bug: webrtc:10125
Change-Id: I659d8f8f3ce3c643710156fa81176ceeaedd714a
Reviewed-on: https://webrtc-review.googlesource.com/c/114165
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26016}
Some tests had to be updated due to this change.
Bug: webrtc:9510
Change-Id: I79c4c0166d8ba5e8190a607d5d35b67dc30a3c14
Reviewed-on: https://webrtc-review.googlesource.com/c/113522
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25952}
The limit we put on probing is a bit too conservative now. If an
allocation limit is set, this CL allows probing up to 2x the current
max allocation limit.
This better handles overshooting when networks actually have the
capacity to allow bursts.
Bug: webrtc:10070
Change-Id: I0003f6b22512c13b6a83c1934952a2c3a2b70b48
Reviewed-on: https://webrtc-review.googlesource.com/c/112905
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25888}
Conversion to kbps will fail if the estimate is lower than the deviation
estimate * 3, since that would produce a negative value.
Bug: webrtc:9718
Change-Id: I83b52acd476d90b1f22c9db9894fa26c9a3e8e17
Reviewed-on: https://webrtc-review.googlesource.com/c/112560
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25854}
This prepares for future refactoring of rate controller.
Bug: webrtc:9718
Change-Id: I425c8c547399bda98b4271a0d24a0bb7ee06bc13
Reviewed-on: https://webrtc-review.googlesource.com/c/112420
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25846}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
The intention is to provide a bandwidth estimate that only updates if
the actual available bandwidth is known to have changed. This will be
used in media streams to avoid changing the configuration (such as
frame size, audio frame length etc), just because the control target
rate changed.
Bug: webrtc:9718
Change-Id: I17ba5a2f9e5bd408a71f89c690d45541655a68e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107726
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25772}
This prepares for providing an additional implementation of delay based
rate control. By moving the probe controller, less code will have to be
added in the upcoming CL.
Bug: webrtc:9718
Change-Id: I64eb2c8f5f7950b6e9d209f110dc0a757c710b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/111860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25770}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
- Rename avg_max_bitrate_kbps to link_capacity_estimate_kbps and change
the type to optional.
- Remove the RateControlRegion enum. The old code seems to have the invariant
that the region is kRcMaxUnknown iff avg_max_bitrate_kbps is uninitialized.
- Change floats to double.
Bug: webrtc:9942
Change-Id: Ic071a11ec4950053ec92beaa06f28f43192521d7
Reviewed-on: https://webrtc-review.googlesource.com/c/111247
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25752}
Test is flaky. See linked bug.
TBR=srte@webrtc.org
Bug: webrtc:10036
Change-Id: I21dd0daceaca6071364cb3aec50da79480f4dfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/111747
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25746}
This won't be perfect since the peeked value will be noisy, but since we
cap it with the starting rate, it should only improve things.
Bug: webrtc:9718
Change-Id: Id2cf42fb85c8d7126f6d538a3982d65caa7a75b7
Reviewed-on: https://webrtc-review.googlesource.com/c/109926
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25604}
This makes it safer to reason about the common case where send
time information is available. We don't have to either assume that
it's available, or check it everywhere the PacketResult struct is used.
To achieve this, a new field is added to TransportPacketsFeedback
and a new interface is introduced to clearly separate which field is
used. A possible followup would be to introduce a separate struct.
That would complicate the signature of ProcessTransportFeedback.
Bug: webrtc:9934
Change-Id: I2b319e4df2b557fbd4de66b812744bca7d91ca15
Reviewed-on: https://webrtc-review.googlesource.com/c/107080
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25465}
Since they rely on a real time simulation, a new build target is
introduced that is intended to be used for real time tests.
Bug: webrtc:9518
Change-Id: Iea58f6a2b687f026e9ab1f37b4aabf8261ed7d23
Reviewed-on: https://webrtc-review.googlesource.com/c/107345
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25410}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
This changes the behavior to a probe only gets trigged if
the total max allocated bitrate actually changed.
Also adding helpful log dump flag to ramp up tests that
was used to investigate the issue.
Bug: chromium:894434
Change-Id: I907675b8fd5a339f838b07d433ecf837e312def1
Reviewed-on: https://webrtc-review.googlesource.com/c/105981
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25212}
Since the pushback controller doesn't strictly adhere to the congestion
window, it better belongs together with the congestion controller logic.
Also ensuring that it does not override the configured min bitrate.
Bug: webrtc:9586
Change-Id: I57dcfc946d470247e66c67adabddaafa3d9d83ad
Reviewed-on: https://webrtc-review.googlesource.com/c/105102
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25115}
This reverts commit 35b5e5f3b0.
Reason for revert: Breaks downstream project
Original change's description:
> Using units in SendSideBandwidthEstimation.
>
> This CL moves SendSideBandwidthEstimation to use the unit types
> DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
>
> Bug: webrtc:9718
> Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
> Reviewed-on: https://webrtc-review.googlesource.com/c/104021
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25029}
TBR=terelius@webrtc.org,srte@webrtc.org
No-Try: True
Bug: webrtc:9718
Change-Id: Iaf470f1eec9911ee6fc7c1b4f5db9675d89d3780
Reviewed-on: https://webrtc-review.googlesource.com/c/104480
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25035}
This CL moves SendSideBandwidthEstimation to use the unit types
DataRate, TimeDelta and Timestamp. This prepares for upcoming changes.
Bug: webrtc:9718
Change-Id: If10e329920dda037b53055ff3352ae7f8d7e32b8
Reviewed-on: https://webrtc-review.googlesource.com/c/104021
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25029}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
Adds SimulatedTimeClient, a class that simulates time so congestion
controllers can be tested using the Scenario test framework without
running in real time.
This allows using simplified scenario tests as unit tests, narrowing
the gap between end to end tests and unit tests.
Bug: webrtc:9510
Change-Id: I61ab388bd610f636b926675b1f14b8d85e3c1114
Reviewed-on: https://webrtc-review.googlesource.com/99801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24890}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
This CL removes some indirection and moves some constants. This
is done to simplify understanding and debugging of the code.
Bug: webrtc:9718
Change-Id: Ibe2b1da0163b4c97ffd1a5bc157f6aa59582d697
Reviewed-on: https://webrtc-review.googlesource.com/98240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24732}
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.
It also adds the possibility to suppress these warnings because
they trigger in a few places.
The long term goal is to avoid regressions on this and remove all the
suppressions.
Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
In-lining GetNetworkParameters and MaybeUpdateCongestionWindow which
was left over from previous refactoring. This prepares for upcoming CLs
changing the behavior.
Bug: webrtc:9586
Change-Id: I6f038acdf97c3db2c85254a36592c617a5754a96
Reviewed-on: https://webrtc-review.googlesource.com/97605
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24570}
This is useful for debugging and has minimal effect on production code.
Bug: webrtc:9510
Change-Id: I3a71f484a0d4e54999e376b7924b73230d86cd96
Reviewed-on: https://webrtc-review.googlesource.com/97607
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24557}
Negative infinity is useful in comparisons as it represents a value
guaranteed to be less than any other timestamp. This avoid using
Timestamp::ms(0) to represent such a value or having to use optional
for the special case of first time execution.
To avoid ambiguity, Timestamp::Infinity is renamed to PlusInfinity.
Bug: webrtc:9709
Change-Id: Ib2debff91036e94d1dc903ec0c6faae30febde4e
Reviewed-on: https://webrtc-review.googlesource.com/79440
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24556}
Sine a starting rate field was added to the constraints struct. Having
it in the initial config separately is reduntant. To simplify the code,
the extra field is removed. This is a follow up on:
https://webrtc-review.googlesource.com/c/src/+/92624
Bug: webrtc:9586
Change-Id: I9b01b16b2fc4b8479e83b7e998308be2295e0325
Reviewed-on: https://webrtc-review.googlesource.com/96801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24514}
Current set of warnings that need fixing before this warning can
be enabled is here -> https://pastebin.com/raw/jTddgPzP
BUG=chromium:588506, webrtc:9451
Change-Id: Id7896cf48c7231b2ee28dde378ff3ce17da73c2b
Reviewed-on: https://webrtc-review.googlesource.com/96623
Commit-Queue: Will Harris <wfh@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24503}
This CL fixes a bug in the feedback based GoogCC where packets lost
was swapped with expected packets received. Since this version of
GoogCC isn't yet used this wasn't discovered. There was also a lack
of unit test coverage. To ensure reasonable behavior, unit tests was
added.
Unit tests was also converted from relevant unit tests on send side
congestion controller for the regular GoogCC controller.
Bug: webrtc:9586
Change-Id: I83c40ff4766104820cb72ec1e8b95c5782def19a
Reviewed-on: https://webrtc-review.googlesource.com/59401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24498}
This adds a start bitrate field to TargetRateConstraints since this is
required in VideoSendStreamTests. This also reduces the differences
between the old SendSideCongestionController and the new TaskQueue
based version.
Bug: webrtc:9586
Change-Id: I5d3d1414e9d30b51723c911a0bf2e96b876c04e5
Reviewed-on: https://webrtc-review.googlesource.com/92624
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24384}
This CL adds a factory for creating a GoogCC network controller that
can be used without RTCP specific messages. This prepares for enabling
use of other underlying protocols as long as they can provide per
packet feedback.
Bug: None
Change-Id: I6671181949d97abd18843d0f4edf75040cc3f007
Reviewed-on: https://webrtc-review.googlesource.com/84583
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24198}
This changes makes the usage of the new probe controller reflect how the
old probe controller was used. That is probes are now sent as soon as
they are generated. This is to avoid regressions in performance doe to
the timing of the sent probes.
Bug: chromium:868776
Change-Id: I722585689258c9b01e8f1dc47249b284a05a2793
Reviewed-on: https://webrtc-review.googlesource.com/91441
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24175}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I289795c92958fd43fed6165894510ad63ca9d24d
Reviewed-on: https://webrtc-review.googlesource.com/90415
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24114}
After detecting overuse of the network capacity, the target
bitrate is reduced. Normally, this should happen at most once
per RTT to prevent repeated reductions from the same overuse
signal. This CL fixes a bug that allowed repeated reductions
if an overuse was detected before it had the first reliable
throughput measurement.
The fix is guarded by a field trial. To enable the fix, use
WebRTC-BweInitialBackOffInterval/Enabled-200/
Bug: webrtc:9493
Change-Id: Iae566227fd94ebb8a4449406572158a8b79d9c53
Reviewed-on: https://webrtc-review.googlesource.com/88765
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24021}
This removes the old version of Probe Controller. The new controller is
slightly different, therefore the legacy SendSideCongestionController is
changed to accommodate the new function.
Most notably, the functionality is changed so that probes are now sent
only from the OnProcess call and not immediately on changing a
parameter.
The lock previously owned and used by ProbeController is moved to SendSideCongestionController. This should not change any
behavior.
Bug: webrtc:8415
Change-Id: I3c69ddeb04aeeae1234a2a5e116fb677f36b4ae4
Reviewed-on: https://webrtc-review.googlesource.com/86541
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23973}
The current acknowledged bitrate (i.e. throughput) estimator takes 500ms
to initialize. This CL creates a field trial to experiment with this
arbitrary initialization time.
Bug: webrtc:9492, webrtc:7746
Change-Id: I8a803f7bc0ee78856e808e289f37bab57d763efa
Reviewed-on: https://webrtc-review.googlesource.com/87145
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23852}
These are useful for plotting creating data files that can be used to
visualize and debug congestion controller behavior.
Bug: webrtc:9467
Change-Id: I75b03a309b4b7d562fefe82a828ae1e6a9f069c8
Reviewed-on: https://webrtc-review.googlesource.com/86126
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23822}
This removes the legacy DelayBasedBwe to reduce code redundancy and
avoid the risk of applying changes on only one version.
Bug: webrtc:8415
Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a
Reviewed-on: https://webrtc-review.googlesource.com/85364
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23798}
This replaces the old AlrDetector used by the pacer with the one in
GoogCC. This reduces the risk of accidentally changing only one version.
Note that the pacer instance will be removed when moving over to the
task queue based send side congestion controller.
Bug: webrtc:8415
Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5
Reviewed-on: https://webrtc-review.googlesource.com/85363
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23791}
Remove unused member noise_var from RateControlInput struct.
Rename incoming_bitrate to estimated_throughput_bps to reflect
that the AimdRateControl may be running on either the send side
or the receive side.
Bug: webrtc:9411
Change-Id: Ie1ae0c29dc9559ef389993144e69fcd41684f1a4
Reviewed-on: https://webrtc-review.googlesource.com/83728
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Anastasia Koloskova <koloskova@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23783}
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.
Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script from modules with parameters
'pacing video_coding congestion_controller remote_bitrate_estimator':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118
Reviewed-on: https://webrtc-review.googlesource.com/83900
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23640}
When CongestionWindowPushback experiment is enabled, the pacer is oblivious to the congestion window. The relation between outstanding data and the congestion window affects encoder allocations directly.
Bug: None
Change-Id: Iaacc1d460d44a4ff2d586934c4f9ceb067109337
Reviewed-on: https://webrtc-review.googlesource.com/74922
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23411}
This CL replaces components in the congestion controller module
that are identical to equivalent components in the rtp and goog_cc
subfolder. Some redundant components are left as they were not
trivial to replace.
Bug: webrtc:8415
Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37
Reviewed-on: https://webrtc-review.googlesource.com/78521
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23384}
This reverts commit 18d7c7ea7e.
Reason for revert:
This seems to cause the auto roller to Chrome to fail on Linux and Mac on the browsertest
WebRtcSimulcastBrowserTest.TestVgaReturnsTwoSimulcastStreams
https://chromium-review.googlesource.com/c/chromium/src/+/1064736
Original change's description:
> Configure and use max bitrate to limit the AIMD controller estimates.
>
> Bug: webrtc:9275
> Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
> Reviewed-on: https://webrtc-review.googlesource.com/77081
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23287}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I8ed827ab6b2f7d2b70b9889e5a88701bfb974d35
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9275
Reviewed-on: https://webrtc-review.googlesource.com/77660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23291}
This prepares for allowing injection of a network controller.
Bug: webrtc:9155
Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3
Reviewed-on: https://webrtc-review.googlesource.com/73100
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23188}
This makes the dependency graph simpler and prepares for moving the
unit classes to api/.
Bug: webrtc:9155
Change-Id: I1b36d5e05f75d70ba8951e880d76359f896f7741
Reviewed-on: https://webrtc-review.googlesource.com/74920
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23161}
This removes places where the units types are implicitly left
uninitialized in network_types.h and adds rtc::Optional where needed.
Also removing the change indicator in the NetworkEstimate struct as it
is not used in practice.
Bug: webrtc:9155
Change-Id: I7e30e338effba96bd466ae91e380e6a8e90f66e1
Reviewed-on: https://webrtc-review.googlesource.com/73369
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23126}
This CL includes the changes from this CL:
https://webrtc-review.googlesource.com/c/src/+/63642
Bug: webrtc:8955
Change-Id: If95cdec59f25e97c6ff5ea45a52d6113128a0921
Reviewed-on: https://webrtc-review.googlesource.com/64822
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22910}
Having only one name for seconds makes the interface more consistent.
The non-abbreviated was chosen since it's used less frequently than
ms() and us().
Bug: None
Change-Id: Ia29ff2f9f18f3dddcde9bac4f041695cef2c8f0f
Reviewed-on: https://webrtc-review.googlesource.com/69817
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22877}
UpdatePacingRates required that a bandwidth estimate was available and
would otherwise crash. This CL ensures that there is an initial bandwidth
estimate available from the beginning.
Bug: webrtc:8415
Change-Id: I20c3b444eac42326a78cfebee70b4c1aa370c867
Reviewed-on: https://webrtc-review.googlesource.com/69802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22857}
The ostream operators does not work consistently with gtest since the
definitions would have to be included before the first include of gtest.
This is tricky to ensure and the end result is that the operators will
sometimes work and sometimes not without obvious explanations.
To avoid causing confusing behavior, this CL removes the operators
pending a better solution.
Bug: None
Change-Id: I66bead0efb7246d368359ddf9e9bfad9d67c05da
Reviewed-on: https://webrtc-review.googlesource.com/68640
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22808}
