With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.
Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.
Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.
Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
This is a reland of 63db77007b that
was broken as I flipped != and == :(
Luckily this made a test flaky, and hence was the original change reverted.
Original change's description:
> Add field trial to base stable target rate on loss based target rate
>
> I.e not the pushback_rate that includes the congestion window pushback
> (if enabled).
>
> Bug: None
> Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30189}
Bug: None
Change-Id: Ia637d0498e6c0c2708eba659e2a30f3235944d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165391
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30196}
This reverts commit 63db77007b.
Reason for revert: Flipped !=which should have been == makes tests
Original change's description:
> Add field trial to base stable target rate on loss based target rate
>
> I.e not the pushback_rate that includes the congestion window pushback
> (if enabled).
>
> Bug: None
> Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30189}
TBR=brandtr@webrtc.org,srte@webrtc.org,jonaso@webrtc.org
Change-Id: I883edb8a74f1ae2a4d783b9825cc08c6a5228aa9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165388
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30193}
I.e not the pushback_rate that includes the congestion window pushback
(if enabled).
Bug: None
Change-Id: I413d011004a95da03dd62f5b423abc3c8b66b333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165383
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30189}
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}
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}
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}
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}
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}
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}
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}
