Commit graph

9 commits

Author SHA1 Message Date
Danil Chapovalov
0040b66ad3 Replace rtc::Optional with absl::optional
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}
2018-06-18 10:24:48 +00:00
Anastasia Koloskova
6d19180030 Fix increase in send rate when not receiving feedback
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}
2018-06-11 16:44:19 +00:00
Per Kjellander
dd3eae5f94 Revert "Configure and use max bitrate to limit the AIMD controller estimates."
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}
2018-05-18 07:12:26 +00:00
Bjorn Terelius
18d7c7ea7e 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}
2018-05-17 16:07:42 +00:00
Oskar Sundbom
18f26d1434 Optional: Use nullopt and implicit construction in /modules/remote_bitrate_estimator
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=terelius@webrtc.org

Bug: None
Change-Id: I1d2538482e2390d91f3285124d011a578da4b61b
Reviewed-on: https://webrtc-review.googlesource.com/23564
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20710}
2017-11-16 12:42:24 +00:00
Ivo Creusen
46ca2879e1 Reland of https://chromium-review.googlesource.com/c/external/webrtc/+/616724 under field trial.
Bug: webrtc:8105
Change-Id: I8c68e0f270b3bd5d8da28b8334d4689064f607f6
Reviewed-on: https://webrtc-review.googlesource.com/4920
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20205}
2017-10-09 11:07:57 +00:00
alexnarest
b335e31bcb This is a rollback of https://chromium-review.googlesource.com/c/external/webrtc/+/616724
it degraded results of the ANA testing

BUG=webrtc:8105

Review-Url: https://codereview.webrtc.org/3011323002
Cr-Commit-Position: refs/heads/master@{#19902}
2017-09-19 19:00:32 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
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_unittest.cc (Browse further)