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}
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 rtc:SentPacket was removed to a separate header. Some usages of
socket.h can be replaced with sent_packet.h which defines a lot less
things, making future maintenance simpler.
Bug: webrtc:9586
Change-Id: If705edda293c389cf2a175117db52a6720a7be86
Reviewed-on: https://webrtc-review.googlesource.com/c/106144
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25201}
This is part of a series of CLs preparing to remove
SendSideCongestionController as a separate class.
Bug: webrtc:9586
Change-Id: I0dabd00793e7b436a679d2ef695d2e557a35ae87
Reviewed-on: https://webrtc-review.googlesource.com/c/105420
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25178}
This moves simple logic from SendSideCongestionController to
TransportFeedbackAdapter. The purpose is to make it easier to
reuse TransportFeedbackAdapter without requiring everything
in SendSideCongestionController.
Bug: webrtc:9586
Change-Id: I35acedd15001d75a06c38ece76868afecd6afa18
Reviewed-on: https://webrtc-review.googlesource.com/c/105106
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25177}
This adds tracking of traffic for streams that are part of bitrate
allocation but without packet feedback to send side congestion
controller.
This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.
Bug: webrtc:9796
Change-Id: I13e994461c26638d76e8f2f115e6d375e4403116
Reviewed-on: https://webrtc-review.googlesource.com/c/104940
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25126}
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}
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.
Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
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 is the first step in order to remove runtime_enabled_features
code from WebRTC.
Bug: webrtc:9693
Change-Id: Ic67f770c2166755ea45c782efb3e4184433ac15e
Reviewed-on: https://webrtc-review.googlesource.com/102361
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24868}
This avoids crashing if an application resets the starting bitrate
before adding streams to the call.
Bug: webrtc:9586
Change-Id: I8d31aba1f4fee40c67c8930f5a32d17700ccadc3
Reviewed-on: https://webrtc-review.googlesource.com/101680
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24808}
Uninline RTPFragmentaion functions
fix RTPFragmentation move constructor and assign operators (was recursive for win)
replace assert with rtc::dchecked_cast
Remove unused includes and dependencies.
Fix other targets that used those includes transitively instead of directly
Bug: None
Change-Id: I647cb1eda107dc7d87d25234095545bc2842fa40
Reviewed-on: https://webrtc-review.googlesource.com/100500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24759}
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 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: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
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 makes calls to send side congestion controller that originates
from the task queue execute directly rather than posting a task. This
ensures that side effects are applied by the time the call returns.
This reduces the risk that the task queue version of the congestion
controller introduces races that does not exist in the process thread
based version.
Bug: webrtc:9586
Change-Id: I82de032dc971c791a0f86d20ccbd47cbb09eba4b
Reviewed-on: https://webrtc-review.googlesource.com/85360
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24382}
Network controller is an implementation of the
NetworkControllerInterface which is a part of congestion controller API.
Bitrate controller computes rate update each iteration (see
https://www.usenix.org/system/files/conference/nsdi18/nsdi18-dong.pdf).
Bug: webrtc:9434
Change-Id: I48d3d9e1c713985ef9ebe28dc1f1285757588c69
Reviewed-on: https://webrtc-review.googlesource.com/87222
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24376}
The PCC congestion control algorithm divides time into consecutive
intervals called monitor intervals. This CL adds a class that is used by
PCC to measure the performance of sending at a certain rate during one
monitor interval.
Bug: webrtc:9434
Change-Id: Ia0447e224067d4ca807bcc6fd8083f9083385b91
Reviewed-on: https://webrtc-review.googlesource.com/87140
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24368}
This is a part of series of CLs adding PCC (Performance-oriented
Congestion Control).
Bug: webrtc:9434
Change-Id: Idd36d8abea008623ac64b939d0de7ee6001f7f23
Reviewed-on: https://webrtc-review.googlesource.com/86951
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24364}
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}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
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}
This CL makes the congestion window parameters, initial window, minimum
window, and maximum window more similar to the values for the
implementation in QUIC.
It also contains minor behavioral changes to better match the Quic
implementation.
Bug: webrtc:8415
Change-Id: I26f4b35b6cbb00178ea47a4aee871b1b700c153b
Reviewed-on: https://webrtc-review.googlesource.com/83587
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23630}
Adds a simple loss rate filter to the BBR network congestion controller.
The loss rate is used to control error correction. Previously the value
was reported as zero which would disable error correction.
Bug: webrtc:8415
Change-Id: Icec8f25fcc9509432ea91eaec30b39a024f92b42
Reviewed-on: https://webrtc-review.googlesource.com/78263
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23467}
A previous change caused a regression in the congestion window behavior.
This CL restores previous behavior.
Bug: webrtc:8415
Change-Id: Id2e42d66bcfb58780c98da2227da39b970f26f0e
Reviewed-on: https://webrtc-review.googlesource.com/79483
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23428}
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I7b7e4769772d67cc5112969fefd4e56c6c72432e
Reviewed-on: https://webrtc-review.googlesource.com/76600
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23419}
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 saves having to iterate trough all packets in flight to compute the
number of outstanding bytes.
Bug: webrtc:8415
Change-Id: I35b135f37649a38b44a36d300af42a815f85192d
Reviewed-on: https://webrtc-review.googlesource.com/77727
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23398}
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 avoid sending more padding than required for the current target
constraints.
Bug: webrt:8415
Change-Id: I3a668990f026414ab78f8406248cde18b81123cc
Reviewed-on: https://webrtc-review.googlesource.com/77763
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23364}
The BBR controller did not properly report updates to congestion
windows. This was due to a check to avoid the overhead of callbacks.
In the current design without callbacks in the controller, the check can
be removed. If helpful for performance, it should live outside of the
controller.
Bug: webrtc:8415
Change-Id: Idf6d6e76fe6d0450841e706019110307e559c11d
Reviewed-on: https://webrtc-review.googlesource.com/78181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23363}
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I6b182246c988dd4a95681c063dcaa779088d0e99
Reviewed-on: https://webrtc-review.googlesource.com/76481
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23356}
The Quic BBR implementation uses packet sequence numbers to keep track
of the time slots used for calculation of send receive rates. To avoid
protocol dependence the port were initially written to use send times
instead.
As there are issues with running BBR in WebRTC, it makes sense to
use an identical implementation as in Quic to ensure that there
aren't implementation issues causing bad behavior. This requires
providing sequence numbers.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: I2cd96bc6ffb88042bb2b91421bfe6cbf7c1ff8ac
Reviewed-on: https://webrtc-review.googlesource.com/76583
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23353}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
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 making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.
Bug: webrtc:8415
Change-Id: Ic2dc4394dc9923e5109ffa5f146c23b527f0c395
Reviewed-on: https://webrtc-review.googlesource.com/76582
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23262}
