Add new histograms
WebRTC.Video.DecodeTimePerFrameInMs.[codec].[resolution].[decoder]
These histograms are more explicit than the existing histogram
WebRTC.VideoDecodTimeMs, since they allow to see performance per
codec/resolution/decoder and also contain per frame statistics instead
of an average decode time.
There's a killswitch, WebRTC-DecodeTimeHistogramsKillSwitch, that can be
used to disable the histograms.
Bug: chromium:1007526
Change-Id: I9f75127b4bc5341e9f406c64ed91164564290b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157881
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29572}
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.
Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
Cname callback is used only on receive side, and statistics (soon)
only on the send side.
Bug: webrtc:10679
Change-Id: I122e9cafaea93cd0ba75dc955a652d9d4bddc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147867
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28767}
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}
Only interesting call deleted in cl
https://codereview.webrtc.org/2704183002.
Move call to QualitySample (used for bad call detection) to
OnRenderedFrame
Bug: webrtc:7408
Change-Id: I0e9ae2ed62fe19a282377cb840e38bd2aae8f3e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128768
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27243}
Unused since https://codereview.webrtc.org/1693553002
Also drop an unneeded include of video_stream_decoder.h.
Bug: webrtc:7408
Change-Id: I249ecfe41b55b59abbd2e880ef144d64f130b0b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128767
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27242}
This change introduces a new measurement into the VideoReceiveStream::Stats
structure to measure the latency between the first frame being received and
the first frame being decoded in WebRTC. The goal here is to measure the latency
difference when a FrameEncryptor is attached and not attached.
Change-Id: I0f0178aff73b66f25dbc6617098033e226da2958
Bug: webrtc:10105
Reviewed-on: https://webrtc-review.googlesource.com/c/113328
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25956}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
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}
This class will be used in upcoming VideoQualiyObserver.
Bug: none
Change-Id: I7d79a6caf3040a3f707ed8700842dea1de81e0a6
Reviewed-on: https://webrtc-review.googlesource.com/77724
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23343}
Reported to UMA and logged for at the end of the call.
Bug: webrtc:8355
Change-Id: I4ef31bf9e55feaba9cf28be5cb4fcfae929c7179
Reviewed-on: https://webrtc-review.googlesource.com/53760
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22132}
Specifically, I'm moving
histogram_percentile_counter.h
mathutils.h
mod_ops.h
moving_max_counter.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I78a999984a27ef935be2d7c3136475d5f209adda
Reviewed-on: https://webrtc-review.googlesource.com/20870
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20832}
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}
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}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
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}
