Commit Graph

12 Commits

Author SHA1 Message Date
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
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: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00
81327d54f3 Move stats for delayed frames to renderer from VCMTiming to ReceiveStatisticsProxy.
Bug: none
Change-Id: If62cc40cf00bc4d657a31a89640d03812cff388e
Reviewed-on: https://webrtc-review.googlesource.com/74500
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23526}
2018-06-07 07:39:37 +00:00
879f5a34a5 Add test against crashes in VideoQualityObserver
If there were a lot of pauses in the receive video stream, it may've
caused a crash because of a null rtc::Optional dereferencing.

This is the test reproducing that behaviour.

Bug: webrtc:9338
Change-Id: I1cef72a88a54f762ef27665d372e4a1d1225e059
Reviewed-on: https://webrtc-review.googlesource.com/80161
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23470}
2018-05-31 12:32:37 +00:00
94150ee487 Implement VideoQualityObserver
This class receives data about video frames from ReceiveStatisticsProxy,
calculates spatial and temporal quality metrics and outputs them to UMA
stats. It is all done in a separate class because it will be further
extended to calculate aggregated quality metrics in the future.

Bug: webrtc:9295
Change-Id: Ie36db83e10c0e8da0b9baa392651cb9a67a54a80
Reviewed-on: https://webrtc-review.googlesource.com/78220
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23387}
2018-05-24 14:53:31 +00:00
b9b07eaf28 Move stats for decoded frames per second from VCMTiming to ReceiveStatisticsProxy.
Bug: none
Change-Id: I631a2b1cb550dded6d1e1daf47ac35583298d30d
Reviewed-on: https://webrtc-review.googlesource.com/36121
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21757}
2018-01-25 09:15:31 +00:00
8e07c134ab Optional: Use nullopt and implicit construction in /video
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.

Bug: None
Change-Id: Ie622c215e06956d8d5629733c76f531b7af45012
Reviewed-on: https://webrtc-review.googlesource.com/23568
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21535}
2018-01-09 15:14:10 +00:00
c7c4191325 Declare the RTCP packets_lost field as signed in the API.
The definition of this field in RFC 3550 says that under certain
conditions it may have a negative value. This change exposes that
property in the WebRTC API.

Bug: webrtc:8626
Change-Id: I4ee249da045dcee940db66ebd915268a97fc13db
Reviewed-on: https://webrtc-review.googlesource.com/31260
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21159}
2017-12-08 11:22:09 +00:00
b06b358207 Update aggregating interval in getStats for receive side.
Now some metrics would be aggregated over 1s window instead of 10s.

Bug: webrtc:8402
Change-Id: If0b5a70257c767a1741b451585f3da501b903374
Reviewed-on: https://webrtc-review.googlesource.com/11400
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20316}
2017-10-16 17:19:07 +00:00
ed23be91b8 Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy.
This should be a separate utility class, rather than internal thing in
ReceiveStatisticsProxy.

Bug: webrtc:8347
Change-Id: I9c8238e625999dba5776d6038c819732d07e9656
Reviewed-on: https://webrtc-review.googlesource.com/7609
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20267}
2017-10-13 08:32:07 +00:00
daa4f7abf5 Calculate and report to UMA 95th percentile of Interframe Delay
Histogram based percentile counter is added in ReceiveStatisticsProxy.
New 95th percentile metric is reported in the same way as interframe
delay.

Bug: webrtc:8347
Change-Id: I5e476cbb6361dd341cdb97c37d883c3923e5f611
Reviewed-on: https://webrtc-review.googlesource.com/6880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20184}
2017-10-06 11:48:34 +00:00
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
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