This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.
Reason for revert: Speculative revert: breaks downstream project
Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
>
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
>
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
>
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
>
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
>
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
>
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}
TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
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}
To avoid incorrect QP sum in the reported stats and to avoid log spam
when switching from a decoder that reports QP to a decoder that does
not report QP.
Bug: None
Change-Id: Ib5ef4d6227344b0d03c3d75596b4a07ef427ae1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155444
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29373}
Number of received FEC bytes is used for the
WebRTC.Video.FecBitrateReceivedInKbps UMA histogram. Before this cl,
that value is based on a FEC packet counter updated by
ReceiveStatistics::FecPacketReceived. This cl deletes that method, and
instead adds a byte count to the FecPacketCounter struct, which is
maintained by the UlpFecReceiver and used for other FEC-related stats.
Bug: webrtc:10917
Change-Id: I24bd494b6909a2fe109d28e2b71ca8f413d05911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150533
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28976}
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}
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 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 change is based on a discussion for integrating a new statistic that
measures the delay between the first frame being received and the first frame
being decoded. To enable this in the context of FrameEncryption it makes sense
for packet receive timestamps to be unconditionally recorded.
Bug: webrtc:10105
Change-Id: I6b3b0118121db1fe5d4a4fb16cf5d94341cd2b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/113487
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25931}
Now WebRTC.Video.MediaBitrateReceived.S0 UMA metric will be counted more
correctly. Before, only keyframes were counted there. Now except some
occasional reorderings near content_type switch, all frames should be
counted correctly.
Note,
WebRTC.Video.MediaBitrateReceived will still be larger than sum of sliced
variants because it includes header overhead while sliced metrics do not.
Bug: none
Change-Id: Ia25d6e3efb572f3fe2e9651996b2243716698140
Reviewed-on: https://webrtc-review.googlesource.com/c/106702
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25253}
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}
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 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}
So that future CLs can de-inline its methods.
We do this by asking the caller to allocate the buffer instead of
having it as a data member.
Bug: webrtc:8982
Change-Id: I246b0973e54510fdd880c3b6875336c31334d008
Reviewed-on: https://webrtc-review.googlesource.com/60000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22355}
This is a fairly minimalistic string building class that
can be used instead of stringstream, which is discouraged
but tempting to use due to its convenient interface and
familiarity for anyone using our logging macros.
As a starter, I'm changing the string building code in
ReceiveStatisticsProxy and SendStatisticsProxy from using
stringstream and using SimpleStringBuilder instead.
In the case of SimpleStringBuilder, there's a single allocation,
it's done on the stack (fast), and minimal code is required for
each concatenation. The developer is responsible for ensuring
that the buffer size is adequate but the class won't overflow
the buffer. In dcheck-enabled builds, a check will go off if
we run out of buffer space.
As part of using SimpleStringBuilder for a small part of
rtc::LogMessage, a few more changes were made:
- SimpleStringBuilder is used for formatting errors instead of ostringstream.
- A new 'noop' state has been introduced for log messages that will be dropped.
- Use a static (singleton) noop ostream object for noop logging messages
instead of building up an actual ostringstream object that will be dropped.
- Add a LogMessageForTest class for better state inspection/testing.
- Fix benign bug in LogTest.Perf, change the test to not use File IO and
always enable it.
- Ensure that minimal work is done for noop messages.
- Remove dependency on rtc::Thread.
- Add tests for the extra_ field, correctly parsed paths and noop handling.
Bug: webrtc:8529, webrtc:4364, webrtc:8933
Change-Id: Ifa258c135135945e4560d9e24315f7d96f784acb
Reviewed-on: https://webrtc-review.googlesource.com/55520
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22203}
Since there's some overhead to each log statement we'll build the entire
log message before logging it.
Bug: webrtc:8529
Change-Id: I04876c7309afdd75985aa84726f8177e5a44bdb5
Reviewed-on: https://webrtc-review.googlesource.com/54301
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22097}
