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}
Store the last known throughput estimate and use that if we're lacking a new measurement.
Bug: webrtc:9363
Change-Id: Ib7a9a495b446bd0f5799382cc095ccff894e0c2b
Reviewed-on: https://webrtc-review.googlesource.com/81749
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@{#23565}
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}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h (Browse further)
