Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.)
With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates.
Bug: webrtc:13402,b/298636540
Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40827}
Start using RobustThoughputEstimator in DelayBasedBwe test in preparation for making it default.
Experiments has not showed significant metric changes. However, simulations has showed that RobustThroughputEstimator better follow the actually receive rate better. Especially during bursts of sent packets. Code is also simpler.
Bug: webrtc:13402 chromium:1411666
Change-Id: I83cfa1fc15486982b18cc22fbd0752ff59c1c1b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40644}
This is to ensure that a bad NetworkState estimate can not decrease BWE
unless an delay BW overuse has been detected.
Bug: webrtc:10489
Change-Id: Ic3a516345999eeba814200c2e295a19b347b2eb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317800
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40628}
Pass FieldTrialsView by const& to note it can't be null and doesn't need to outlive the constructor
In unittests use AimdRateControl object directly instead of through helpers
Use unit types (TimeDelta, DataRate) directly, reducing their conversion to plain numbers
Replace SimulatedClock with a single Timestamp now variable or constant
Bug: None
Change-Id: I147f85e629b4d8923aa19896ea211a6f9dca1e68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299260
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39707}
Removed old disabled tests
enable test on android
Bug: webrtc:4711
Change-Id: Ic9adbdadc9e847bdf31b8be4ce116a3695499944
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284922
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38736}
Replace helper functions with the constant
Remove option to set min bitrate in RemoteBitrateEstimator as unused:
ReceivedSideCongestionController is the only user of the
RemoteBitrateEstimator interface, and it always sets the same value
right after construction that RemoteBitreateEstimators already use.
Bug: None
Change-Id: If179fdd72b1ded6ad1fd0a6dfffc97b302153322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269383
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37613}
This cl adds a new field trial parameter to WebRTC-Bwe-EstimateBoundedIncrease that allows the delay based estimate to immediately increase to the upper link capacity estimate.
Bug: none
Change-Id: I875121a41f65cc8e76bb87bbf421731ba6b4551b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257142
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36392}
This reverts commit 865d94e452.
First patch is the same as original cl. Second patch includes a fix to ensure the clamped bitrate does not increase to 85% of the last network estimate.
Bug: none
Change-Id: Idf1b2af3fb60c0d392c48c1b6c0d8526f900f9d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256016
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36270}
This reverts commit e66e6a845b.
Reason for revert:
May cause BWE to increase on delay increase if link capacity estimate is too high.
Original change's description:
> Apply lower bound of delay based estimate in AimdRateControl::ClampBitrate
>
> This move the functionality of applying the lower bound of a network estimate to AimdRateControl::ClampBitrate instead of ChangeBitrate.
> The purpose is to be able to also clamp probe estimates set by AimdRateControl::SetEstimate as well.
>
> Bug: none
> Change-Id: I6a4d64d2e98bb99da06010e2edaf20dc42880e37
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255823
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36219}
Bug: none
Change-Id: I8c65b1461160dbf3d35e50ef2cc6f9bc305c2b15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256011
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36248}
This move the functionality of applying the lower bound of a network estimate to AimdRateControl::ClampBitrate instead of ChangeBitrate.
The purpose is to be able to also clamp probe estimates set by AimdRateControl::SetEstimate as well.
Bug: none
Change-Id: I6a4d64d2e98bb99da06010e2edaf20dc42880e37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255823
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36219}
This adds a new paramater to WebRTC-Bwe-EstimateBoundedIncrease that ensure that even if the link capacity has decreased, the delay based estimate does not immediately decrease unless an overuse has been detected.
This is a follow up to https://webrtc-review.googlesource.com/c/src/+/252442/
Bug: none
Change-Id: I98d77ba1e3f7856b06f2691575f2d248a500e659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253901
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36154}
capacity.
Allow delay based estimate to increase up to 85% of the current NetworkStateEstimate
even if in ALR. The estimate may not increase higher than that.
WebRTC-Bwe-EstimateBoundedIncrease/ratio:0.85,ignore_acked:true
Bug: none
Change-Id: I6f34af7fab03082ca168e624ddea06f216790fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252442
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36096}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).
Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
CL partially auto-generated with:
git grep -l "\bassert(" | grep "\.[c|h]" | \
xargs sed -i 's/\bassert(/RTC_DCHECK(/g'
And with:
git grep -l "RTC_DCHECK(false)" | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'
With some manual changes to include "rtc_base/checks.h" where
needed.
A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.
The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.
This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).
Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
Delete unused macros BWE_MIN and BWE_MAX.
Move enum RateControlState: Make it a private enum class in
AimdRateControl, the only user.
Change users of the header file that only need BandwidthUsage, to
instead include api/network_state_predictor.h, the file defining this
class. As a result, fewer dependencies on
modules/remote_bitrate_estimator.
Bug: None
Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32704}
This fixes an issue where the delay based target bitrate would increase
unlimited when the WebRTC-DontIncreaseDelayBasedBweInAlr is set.
Bug: webrtc:10542
Change-Id: I1aaf0835a91efc27e95198812b6833dbc24a1485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169843
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30693}
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}
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}
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}
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}
- 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}
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 CL removes some indirection This is done to simplify
understanding and debugging of the code.
Bug: webrtc:9718
Change-Id: I48974d161213b9ef8fc5912bd3dc3f9d85ddfa66
Reviewed-on: https://webrtc-review.googlesource.com/100302
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24730}
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}
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}
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}
