This reverts commit 60ec3703cd.
Reason for revert: Needs back-end test before always enabling.
Original change's description:
> Ensure loss-based controller is always enabled.
>
> The new default parameters are the ones that were used in the Chrome
> Finch trial. The deleted unit test is invalidated by these changes.
>
> Bug: chromium:941413
> Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30049}
TBR=srte@webrtc.org,crodbro@webrtc.org,jonasolsson@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:941413
Change-Id: I5da4676ad8be2569ad7eed99e954e0d0b624110b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161902
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30061}
The new default parameters are the ones that were used in the Chrome
Finch trial. The deleted unit test is invalidated by these changes.
Bug: chromium:941413
Change-Id: I597f4b0defaebe5bb3a6710b071fae2ee5c6f461
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160652
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30049}
Change #159711 adds the option to filter out small packets on the
input to the delay-based BWE. This change adds similar functionality
to BitrateEstimator by reducing the weight of small observations.
Bug: webrtc:10932
Change-Id: I0a673a067f7ef86769cabd30443e60e9de70053c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160009
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29865}
Some of the field trial default values are changed as well.
Now available bitrate estimation will be decreasing when RTT is more than 3 seconds.
Unless different parameters for the field trial are specified.
Bug: None
Change-Id: Icd1923fc2e2e7766a7f645016c5432a52537145f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158840
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Nikita Zetilov <zetilovn@google.com>
Cr-Commit-Position: refs/heads/master@{#29823}
crodbro@ found that the previous field trial, which filtered the deltas
in the trendline estimator, can increase the noise caused by varying
packet sizes. Moving the filtering to the DelayBasedBwe class fixes the
issue.
To avoid confusion, we've updated the field trial name, so e.g.
WebRTC-BweIgnoreSmallPacketsFix/small:200bytes,large:200bytes,
fraction_large:0.25,smoothing:0.1/
should be used to enable the feature.
Bug: webrtc:10932
Change-Id: If77e83043c37fff909038405f634e541ce41abb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159711
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29804}
Add interface for AcknowledgedBitrateEstimator
Add simplified throughput estimator, implementing the same interface.
The choice of estimator implementation can be controlled by a field trial.
Bug: webrtc:10274
Change-Id: I6bef090a8a6a1783f3f5750a2ee56189f562a9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158892
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29761}
* Removes legacy defines from rtp_rtcp_defines.
* Simplifies the feedback adaptation logic, this is achieved
by using the ability to preserve lost packets information
from the RTCP message.
* Extracts in flight data tracking to a separate helper class.
* Removes legacy fields and constructors from the PacketFeedback
structure.
* Removes the legacy GetTransportFeedbackVector method.
Apart from reducing total LOC, this prepares for moving the adaptation
to run on a TaskQueue.
Bug: webrtc:9883
Change-Id: I5ef4eace0948f119f283cd71dc2b8d0954a1449b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29674}
Removes all unused features, reducing the exposed interface surface.
This makes refactoring and maintenance simpler as we can change
TransportFeedbackAdapter without making corresponding changes
to RtpVideoSender.
Bug: webrtc:9883
Change-Id: If372a868e0765e94df52b4de52d3bb619ce11471
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156943
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29649}
The code path it calls is no longer actually used and will be cleaned
up in a follow-up CL.
This prepares for simplifying the transport feedback adapter and moving
it to run on a task queue.
Bug: webrtc:9883
Change-Id: I750398069414ffa782067d021c0a3837049d98eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158621
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29638}
The feature is added as part a new field trial WebRTC-Bwe-IgnoreProbesLowerThanNetworkStateEstimate
Bug: webrtc:10498
Change-Id: I72b3c73256a35e0583f4d595edef45848f8bbb22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158260
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29624}
It was intended to be used for to fall back to probe rate if ack rate is missing.
This partly reverts commit aa4f100225.
Reason for revert:
Code is unused 1 year after submitted.
Original change's description:
> Adds trial to fall back to probe rate if ack rate is missing.
>
> Bug: webrtc:9718
> Change-Id: I7b6e1d3c051e67b97f6de1ec95e84631af9c5b0d
> Reviewed-on: https://webrtc-review.googlesource.com/c/113600
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25953}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9718
Change-Id: I06804782c2e210d1c484426e915e4d8447572739
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158084
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29597}
The minimum bitrate was lower bounded by the previous value and could thus not become lower when a stream allocation was removed.
Bug: None
Change-Id: I60068dbc7691121f001cbb233ca4a25269047f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157424
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29541}
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}
Corresponding mock class is deleted rather than updated,
since it appears unused.
Bug: webrtc:8422
Change-Id: If1c6c5ed73abff0d2545e8666c4bb8b63ee5b53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/13862
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29505}
They were already tightly coupled, merging them makes the relations clearer.
We also remove the kill switch for removing duplicate feedback events since
there has been no need to use it.
The potential to account for bytes sent in AddNewPacket was also removed
since it is not used by TransportFeedbackAdapter.
Bug: webrtc:9883
Change-Id: I51823e0ce838c22158637954749310e0d0eeff27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156140
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29449}
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}
Renaming and splitting it into helper methods. This is to more clearly
separate the things it does and prepares for moving things to GoogCC.
Additionally, replacing calls with current_target_ as input with
ApplyTargetLimits to better reflect the intended behavior.
Bug: webrtc:9883
Change-Id: I2c47ec74a9cbc271aff91645c763373297f26acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154425
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29346}
This makes them better reflect their contents and usage. Also replacing
zero with infinity where it's used to reflect the lack of a limit.
Bug: webrtc:9883
Change-Id: Ibc498aa3a41d34c16d363e892a927e482949ab51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154423
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29313}
Removing the redundant last_estimated_bitrate_bps_ and renaming some
members to better reflect the contents. Also replacing the CurrentEstimate
method of SendSideBandwidthEstimation with value specific access methods.
Bug: webrtc:9883
Change-Id: I73cb08e09374adddf5991cb3793fa4a4fee20c85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154351
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29304}
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}
In DelayBasedBwe, in experiment WebRTC-Bwe-AlrLimitedBackoff, back off relative the BWE only after the first detected overuse. The first time overuse is detected, back down to the acked bitrate.
The idea is to faster drop BWE in the beginning of the call when the initial BWE guess may be too high. Withouth this, it may take a too long time to initially back down.
BUG=webrtc:10542
Change-Id: I2a11457d2391ad25658e7c13d9cae02a38973ecb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152541
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29163}
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}
This behavior seems to conform to expectations from the rate allocators,
using this signal to chose which layers to enable and then distributing
the remaining bandwidth to the activated layers.
Bug: webrtc:10126
Change-Id: If0e1b27dc672ec2fbb30a5f5ac734e5ed4b42e45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151306
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29065}
The change is behind a field trial. The intention is to use this
to (heuristically) base the bandwidth estimate only on video packets
even if both audio and video packets have transport sequence numbers.
Bug: webrtc:10932
Change-Id: I6cc5bb9ab6f1a3f25b84ee6ac78e4abb4112032e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150787
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29031}
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}
In https://webrtc-review.googlesource.com/c/src/+/138275
the congestion window was recalculated during OnProcessInterval, as
to consider the case when downlink is down. However, this update
was not propagated to the congestion window pusback controller,
nor returned in the update.
This patch fixes that issue, as well as adding two tests to ensure
the behaviour works as expected.
Bug: None
Change-Id: Ic126d929dc7a7a3393a2f34a4682eea1ee1f2240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146704
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28667}
It's not currently used and it complicates receive side estimation.
Bug: webrtc:10742
Change-Id: Iaa3c86807c7b637aea3ff393e728dc91eac23db6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145724
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28623}
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 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}
All usages compare the budget usage to ratios, so we can skip a few
multiplications.
Bug: webrtc:10719
Change-Id: I0205d74762043d972c087c152915e4fdd9510057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140289
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28190}
The probe configuration is currently a single field trial. To allow
multiple experiments with non-overlapping subsets of these keys I've
added a few extra keys that override different subsets of the config.
Bug: webrtc:10394
Change-Id: I54ffd1105129794fcdae4cce314910acaa4074af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138274
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28076}
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}
When backing down, the delay based estimator can still use acked bitrate instead of the last set estimate.
Original code was reviewed in:
https://webrtc-review.googlesource.com/c/src/+/113880
BUG=webrtc:10144
Change-Id: Ia6e2d6d7d05f88f7e51d61b6e37c61a89adccf8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135950
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27912}
This is a reland of 0870c70b04
Original change's description:
> Reland "Improving robustness of feedback matching code in event log parser."
>
> This is a reland of a1e4fbb253
>
> Original change's description:
> > Improving robustness of feedback matching code in event log parser.
> >
> > Removes the dependency on TransportFeedbackAdapter thereby removing
> > some of the complexity that came with it, in particular, we don't fill
> > in missing packets. This makes the code easier to debug and avoids some
> > confusing logging that's not relevant for the parser.
> >
> > Bug: webrtc:9883
> > Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27739}
>
> Bug: webrtc:9883
> Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27763}
Bug: webrtc:9883
Change-Id: I1f80ed1f63ad75fbb97f5f401fe486d19c057f75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134462
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27829}
This reverts commit 0870c70b04.
Reason for revert: Failed to handle lost packets.
Original change's description:
> Reland "Improving robustness of feedback matching code in event log parser."
>
> This is a reland of a1e4fbb253
>
> Original change's description:
> > Improving robustness of feedback matching code in event log parser.
> >
> > Removes the dependency on TransportFeedbackAdapter thereby removing
> > some of the complexity that came with it, in particular, we don't fill
> > in missing packets. This makes the code easier to debug and avoids some
> > confusing logging that's not relevant for the parser.
> >
> > Bug: webrtc:9883
> > Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27739}
>
> Bug: webrtc:9883
> Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27763}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9883
Change-Id: Ibcfc4f7425fe202d86f0c3a33de51e605dc17c04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134312
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27790}
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}
This is a reland of a1e4fbb253
Original change's description:
> Improving robustness of feedback matching code in event log parser.
>
> Removes the dependency on TransportFeedbackAdapter thereby removing
> some of the complexity that came with it, in particular, we don't fill
> in missing packets. This makes the code easier to debug and avoids some
> confusing logging that's not relevant for the parser.
>
> Bug: webrtc:9883
> Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27739}
Bug: webrtc:9883
Change-Id: I460d0c576626614fb4ce2c3d5e3ddbb5d1c122cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134106
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27763}
This reverts commit a1e4fbb253.
Reason for revert: Breaks downstream.
Original change's description:
> Improving robustness of feedback matching code in event log parser.
>
> Removes the dependency on TransportFeedbackAdapter thereby removing
> some of the complexity that came with it, in particular, we don't fill
> in missing packets. This makes the code easier to debug and avoids some
> confusing logging that's not relevant for the parser.
>
> Bug: webrtc:9883
> Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27739}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Icdf3231f5a32b6f63a903c7dffc8ca505680a72a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134105
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27742}
Removes the dependency on TransportFeedbackAdapter thereby removing
some of the complexity that came with it, in particular, we don't fill
in missing packets. This makes the code easier to debug and avoids some
confusing logging that's not relevant for the parser.
Bug: webrtc:9883
Change-Id: I6df8425e8ab410514727c51a5e8d4981d6561f03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133347
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27739}
With this change, both the normal RTP and the transport-wide sequence
numbers are propagated with with AddPacket() call via a new
RtpPacketSendInfo struct, replacing the previous set of parameters.
The intent with this is that SendTimeHistory can hold a mapping from
transport-wide to rtp sequence numbers, and then via callbacks let the
RTP modules know when packets have been received by the remote end.
Bug: webrtc:8975
Change-Id: I6a24fc6282cbb041393752d39593c2867b242192
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133021
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27708}
The idea is that when ALR is detected, the encoder can not produce the bitrate
needed for the delay based estimator to detect overuse and thus the delay based
estimator should not be allowed to increase further.
Likewise, if ALR is not detected, the delay based estimator is allowed to
increase the BWE to ensure that there is no region where the BWE can get stuck.
BUG=webrtc:10542
Change-Id: Ic94b708461c9077fd09132ee4ecb6279ffcd5f99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133190
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27661}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig
BUG=webrtc:10335
Change-Id: Ie148cb466f86d8fa1ded5c7f125fbcccf6e7dbe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132714
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27642}
Instead use WebRtcKeyValueConfig and FieldTrialBasedConfig.
The purpose is to allow a user of GoogCC to use different settings on different instances.
BUG=webrtc:10335
Change-Id: I2f837688c9fdd341eecb44484cc784b1c80da1a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132791
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27617}
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.
Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
if currently sent bitrate is application-limited.
Bug: chromium:951299
Change-Id: Ibc1ebd74eaa4a019dc290c11b606796c5be21d0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131126
Commit-Queue: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27539}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
A reasonable amount of incoming packets could generate feedback
for millions of packets.
Bug: chromium:949020
Change-Id: I7f3e6b75b683af5b2732c472cc92c6788540486b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131333
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27481}
The has some benefits:
* We no longer need locks to protect the emulated network node state.
* We only process when there are packets in flight.
* It makes Scenario more similar to network emulation manager.
Bug: webrtc:10365
Change-Id: I8bd1ad1edfb54b047e8109dabd9846ae451cef17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127548
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27393}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
When a bandwidth decrease to the estimated throughput would lead to
the "critical low" region we allow dropping to the link capacity
estimate instead (if it is higher).
Also moved BweInitialBackOffInterval config to the same field trial
string.
Bug: webrtc:10462
Change-Id: I4d6ee020a9ab8cede035b64253e3b3b1e2fb92b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27325}
Problem seems to be that once the estimate drops, "sample_uncertainty"
becomes very large, and it therefore takes a long time to recover.
Fix is under config for further downstream verification.
Bug: webrtc:10462
Change-Id: I5c2035f06e8a5088db0f0cb6ca511ef900e07645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128902
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27320}
This CL allows us to control how many probes we send when the bandwidth
allocation is updated, and how big they are.
Bug: webrtc:10394
Change-Id: I19e40740a528f83384b65d7509295034cc9a3031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129904
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27317}
This lets us change how many bytes and packets goes into the probes, as
well as some other things.
Bug: webrtc:10394
Change-Id: I26bb26a644e6f00366e9275228760c8744d63735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128424
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27312}
This reverts commit 2c7964832e.
NOTE: Build file changes had to be manually reverted to avoid
merge conflict.
Reason for revert: Bad interaction with Chromium issue.
Original change's description:
> Remove rtc::TimeMillis() call from ALR detector.
>
> We want to avoid system clock dependencies in congestion
> controllers as it makes it harder to test them. This CL removes
> a rtc::TimeMillis() call from the AlrDetector class and removes
> dependencies on rtc_base_approved as it exposes time_utils.h.
>
> Bug: None
> Change-Id: Ie50a27399c05a0c50cdc17ad142db884b94ee918
> Reviewed-on: https://webrtc-review.googlesource.com/c/124491
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26879}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:942752
Change-Id: I7fc4391f16779ebb5d3c72a058fc72a3e4c64bce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129440
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27267}
Current implementation of loss based controller has a sensitive filter.
This CL increases the moderate loss rate to ensure robustness to small
changes in network behavior.
Bug: webrtc:10365
Change-Id: I0dcb5ba45904d8dda4c78b39bd13619523bc90ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127901
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27160}
This reverts commit a594ef0893.
Reason for revert: This log is triggered more than 10,000 times per run, spamming the log output to the extent that tests start failing with EXCESSIVE_OUTPUT.
The tests are chromium.webrtc.fyi tests:
* WebRtcStressResolutionSwitchBrowserTest.MANUAL_SurvivesPeerConnectionResolutionSwitching
* WebRtcStressPauseBrowserTest.MANUAL_SurvivesPeerConnectionVideoPausePlaying
on linux, win, and mac.
Example run: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/2556
Original change's description:
> Log an error if the RTT is negative
>
> Bug: webrtc:10407
> Change-Id: I5479cb2b7163c6e9e58854f4ffa7976b3d606da5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127568
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27122}
TBR=srte@webrtc.org,eshr@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10407
Change-Id: Ida2572b722b92bae4893d4567597dd21d1df54b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128120
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27144}
This CL adds enough simulated time to recover the built up delay. This
makes the test less sensitive to small timing changes. This prepares
for further changes in Scenario test framework.
Bug: webrtc:10365
Change-Id: Iddbe6a57e31f17f590004e29221f907321cbb3d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127567
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27107}
This CL adds a field trial that lets us control the size of the initial probes, how we grow the following probes and how big and frequent our ALR probes are.
Bug: webrtc:10394
Change-Id: I6c7783dfada9aaf55cd836dd8991bb7b8ca4993b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27077}
This fixes an issue where the time between freezes dropped in
perf tests. This was triggered by resetting and updating the bitrates
immediately if the min allocatable bitrate changed, causing a drop in
target bitrate. With this CL, the change in min bitrate will not take
effect until we get more data.
Bug: chromium:940349
Change-Id: Ia680a5f1cfe71847ef90669987e7b89b240b9524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126625
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27054}
This prepares for upcoming CL using the value for more than
controlling pacing rates.
Bug: webrtc:9887
Change-Id: Id3891c3727865149b87f946b3e7c3095a6ac9f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126001
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27004}
Since there is no way to enable/disable these diagnostics at runtime,
this CL moves the suppression into the rtc_* templates in order to
remove the need to explicitly add the snippet of code needed to
suppress it (currently copy/pasted in 144 locations).
The diagnostic that causes the most problems is the one about "complex
class/struct explicit ctor/dtor" [1] because WebRTC doesn't find
it useful enough.
Other diagnostics are good (for example the one that warns about
using "virtual" instead of "override", but that will be covered by
this clang-tidy check [2]) while others are Chromium related so
they have never triggered.
[1] - https://cs.chromium.org/chromium/src/tools/clang/plugins/FindBadConstructsConsumer.cpp?l=147-167&rcl=b4bebe1aa15dba7ca5fcc6456a81a55665327c3a
[2] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:163
Change-Id: Icbf27efa5b369100a31e6a32df1a0913729b3b34
Reviewed-on: https://webrtc-review.googlesource.com/c/125088
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26918}
We want to avoid system clock dependencies in congestion
controllers as it makes it harder to test them. This CL removes
a rtc::TimeMillis() call from the AlrDetector class and removes
dependencies on rtc_base_approved as it exposes time_utils.h.
Bug: None
Change-Id: Ie50a27399c05a0c50cdc17ad142db884b94ee918
Reviewed-on: https://webrtc-review.googlesource.com/c/124491
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26879}
In a previous refactor, the ALR probe timestamp update was moved
after a return statement by accident. This CL fixes this.
The impact of this bug is limited as there are several other criteria
that has to be fulfilled for sending ALR probes.
Bug: None
Change-Id: Ia85e6ff9d782c1c4722a3df7e01ed803cf86b11d
Reviewed-on: https://webrtc-review.googlesource.com/c/124489
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26861}
There are two RTT values reported to GoogCC. They come from the same
source initially but one is calculated and smoothed in the video call stats.
However, there's not really any technical reasons why this value should
be received via the stats, this has just been maintained for legacy reasons.
Experiments shows no real difference between the modes, therefore the
stats-reported RTT is removed in this CL as a cleanup.
Bug: None
Change-Id: If1462d6c91570ffb883ecef2ba034f04a571c9b5
Reviewed-on: https://webrtc-review.googlesource.com/c/123883
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26833}
Originally RtcEventProbeClusterCreated was logged in bitrate prober. This means that anyone who was using GoogCcNetworkControl wasn't logging it, and the NetworkControl wasn't self-contained.
This changes moves the responsibility for logging ProbeClusterCreated to ProbeController (where the probe is created), it also moves the responsibility for assigning probe ids to the probe controller.
Bug: None
Change-Id: If0433cc6d311b5483ea3980749b03ddbcd2bf041
Reviewed-on: https://webrtc-review.googlesource.com/c/122927
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26713}
This will be used to investigate the effect of congestion window
pushback on bandwidth esimation. There is currently no data available
in event logs to analyze this in test runs.
Bug: None
Change-Id: I2397842e90fd4acab6306b03d1ee9daf62469ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/121765
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#26681}
With the current implementation, whenever we are toggling between
sending/not sending retransmissions, the BitrateAllocator will
toggle the total requested max bitrate that is signalled to the
probing mechanism. The result is that spurious probes are sent
mid-call, at |max_bitrate| and |2*max_bitrate|. This behaviour
is undesirable, and thus removed in this CL. Instead, whenever
the allocation limits actually change, we produce a single
set of probes at |max_bitrate| and |2*max_bitrate|.
This CL does not change how the BitrateAllocator hysteresis is
accounting for protection, since it does not relate to the
spurious probes.
Bug: webrtc:10275
TBR: sprang@webrtc.org
Change-Id: Iab3a690a500372c74772a8ad6217fb838af15ade
Reviewed-on: https://webrtc-review.googlesource.com/c/120808
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26544}
Calculate packet exit times "just in time" rather than at send time.
This allows changing bandwidth with packets in the queue being reflected
correctly.
Bug: webrtc:10265
Change-Id: I5a38663def4d2bfee64164f9ae62bc61277064bb
Reviewed-on: https://webrtc-review.googlesource.com/c/120403
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26487}
Will focus on delivering model based controller instead.
Bug: webrtc:9718
Change-Id: I5df82424469c577f3c170758e0db64e3e1aa7705
Reviewed-on: https://webrtc-review.googlesource.com/c/120607
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26478}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
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}
Instead timestamps required for processing are provided explicitly.
This makes it easier to ensure correct usage in log processing
and simulation.
Bug: webrtc:10170
Change-Id: I724a6b9b94e83caa22b8e43b63ef4e6b46138e6a
Reviewed-on: https://webrtc-review.googlesource.com/c/118702
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26339}
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}
The BBR controller can still be injected, but the trials
will no longer work. This reduces the binary size.
Bug: webrtc:8415
Change-Id: I2c32c414d08ef0cc16bfd72651535a755cde9916
Reviewed-on: https://webrtc-review.googlesource.com/c/114120
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26077}
This will be used to calculate a lower bound for the round trip time in
a later CL.
Bug: webrtc:9718
Change-Id: I0a1d22045961fe6bd343d1d6ce9b36490b036bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/114680
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26050}
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}
For somewhat similar funtionality, GoogCcNetworkController can
be used via GoogCcNetworkControllerFactory.
Bug: webrtc:9586
Change-Id: I298050184513f50c1b9ef5c21b8c9b7a6ca46fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/114543
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26040}
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}
This CL simplifies a lot of code that can be cleaned up after the merge
of RtpTransportControllerSend and SendSideCongestionController.
In particular, the role of CongestionControlHandler is reduced to only
handle the pacer pushback and stream pausing mechanism.
Bug: webrtc:9586
Change-Id: Idbc1e968efd35e6df6129bc307f6bc1db18d20f2
Reviewed-on: https://webrtc-review.googlesource.com/c/113947
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25994}
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}
This fixes an issue where SendSideCongestionControllerTest.OldFeedback
calls a function that posts a task on a TaskQueue and immediately after
changes the mocked observer that is called from that task.
Bug: webrtc:10056
Change-Id: Ib1cca5bf695482e75106bfc715662e4f76c381d9
Reviewed-on: https://webrtc-review.googlesource.com/c/112940
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25882}
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}
