Old way to produce this histogram was based on RtcpStatisticsCallback
reporting sent RTCP messages, with some additional processing by the
ReportBlockStats class. After this cl, to grand average fraction loss
is computed by StreamStatistician, queried by VideoReceiveStream when
the stream is closed down, and passed to ReceiveStatisticsProxy which
produces histograms.
This is a preparation for deleting the RtcpStatisticsCallback from
ReceiveStatistics.
Bug: webrtc:10679
Change-Id: Ie37062c1ae590fd92d3bd0f94c510e135ab93e8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147722
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28747}
This adds the following non-standardized metrics to video receiver
stats:
- freezeCount
- pauseCount
- totalFreezesDuration
- totalPausesDuration
- totalFramesDuration
- sumOfSquaredFrameDurations
For description of these metrics see
https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
Bug: webrtc:10145
Change-Id: I4c76d5651102e73b1592ffed561e6224f2badeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/114840
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26523}
This adds calculation and reporting of harmonic frame rate (HFR) metric.
HFR is calculated as call_duration_secs / sum(frame_duration_secs ^ 2).
It penalizes long freezes and could better represent user experience
related to smoothness of playback.
Bug: none
Change-Id: I4d2d46deaa44bb4221b53969a1c0a334e0c1bde9
Reviewed-on: https://webrtc-review.googlesource.com/c/117661
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26279}
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}
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}
