This CL also slightly refactors unit test, to test fewer things each.
Bug: webrtc:11508
Change-Id: I98553d2b185364132c626d373683f38891e36c6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31087}
This was introduced on trial but turned out to perform badly for WebRTC
purposes and never used in production.
Bug: webrtc:9883
Change-Id: Ib72acddf4d90fc9ac042084dddf526c04661f290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173680
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31085}
https://webrtc-review.googlesource.com/c/src/+/172847
------------ original 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: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
This is a reland of 12e2d4ddb235da6ec7a5c1c3a83ac33d394920b0
Original change's description:
> APM: Remove the usage of AudioFrame in the AudioProcessing interface
>
> This CL removes the AudioFrame-based APIs from the AudioProcessing
> interface.
>
> Bug: webrtc:5298
> Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31016}
Bug: webrtc:5298
Change-Id: I70e6d59afc3716ee6109d8b9dc384abc71c93624
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173476
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31066}
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}
This CL removes the AudioFrame-based APIs from the AudioProcessing
interface.
Bug: webrtc:5298
Change-Id: Iab470b26b10e06dcf29c543851ae0085bc5b66f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172939
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31016}
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}
Extract test peer creation into separate file to simplify code and
increase readability. Also it is 1st step in bigger refactoring of PC
level test fixture implementation to make it more granular and reusable.
Change-Id: I687a17bda33a8eebc1ef0ddc0d54572e095fd709
Bug: webrtc:11479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172628
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30980}
Since the flag is now on by default, we can remove it (after all
callers stop passing it).
We can also remove all Chart JSON code from WebRTC since it is
no longer used.
Requires one recipe CL and one downstream CL to land first.
Bug: chromium:1029452
Change-Id: Ic1d62e8ab9dfcd255cd2bf51d153db80d59c564b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171878
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30927}
The proto code is copied into the out dir, so always use that since
it is what isolate is using. Previously we pointed straight at the
checkout code.
I think copying python into the out dir is probably the right way
to do things, so we should go that way in the future.
Bug: chromium:1029452
Change-Id: I701cc84a674021d2f78c73db8808f55cd6ae5174
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171877
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30923}
This reverts commit 7b201012bc7f7cd95249b8314d0d7ebabe966d8b.
Reason for revert: Seems to work, but need to get low bw tests working first
Original change's description:
> Flip histograms to true by default, fix unit in isac_fix_test.
>
> Requires downstream changes for all WebRTC perf tests, and
> a corresponding recipe change so isac_fix_test starts using the new
> flow.
>
> Bug: chromium:1029452
> Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30906}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org
Change-Id: I96c2309cd71be14c5a27b515736a32f1b256453c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029452
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171865
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30913}
Requires downstream changes for all WebRTC perf tests, and
a corresponding recipe change so isac_fix_test starts using the new
flow.
Bug: chromium:1029452
Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30906}
To migrate on new GetStats API and properly support target encode bitrate
for regular, simulcast and svc cases we need to calculate it inside video
quality analyzer getting values from SetRates in VideoEncoder.
Bug: webrtc:11381
Change-Id: Ia37acac764ed3c30f64cdbfda8906d543fa03ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171501
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30881}
Since we're now calling the shots of what flags get passed in the
recipes, we can just pass the right ones right away and remove all
the flag renaming.
--isolated-script-test-output is no longer passed, so we can just
remove it. The recipe is currently passing
--isolated-script-perf-test-output but I will start passing the
underscore version shortly.
Bug: chromium:1051927
Change-Id: I571090e62f79ea17c793295df7f5abb21f45d207
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30878}
Use real video duration instead of test duration to calculate harmonic
frame rate in DefaultVideoQualityAnalyzer.
Bug: webrtc:11445
Change-Id: Ia5f96b2f87178419ec6ebe2ff5dbcb5a0c03c824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171104
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30854}
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay
(previously it was "only" network_id)
The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.
OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/
BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
It's better to set this to the right value in the C++ API rather
than the hack in catapult_uploader.py.
Bug: chromium:1029452
Change-Id: Ia942ff22f8422874cd226e6a7fdce20333ac4a50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170632
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30817}
This is a reland of 30026214b13535a9fe1c47f1463378fbf976c643
Original change's description:
> Fix for out-of-bounds write in square test frame generator.
>
> The length is set on construction and includes an assumption on the
> image resolution, if the resolution changes, a square might be larger
> than what fits into the buffer, causing an out of bounds write. This
> CL fixes this simply by restricting the size of the square.
>
> Bug: webrtc:11415
> Change-Id: Iee14a1971997b4ae2fddef0a7af7c76a2509e879
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170042
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30732}
Bug: webrtc:11415
Change-Id: I0dc584858208f478434ebc6f9e31634595c4e5ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170116
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30779}
We need to write protos as "wb" and not "w", otherwise we get CRLF
on Windows which corrupts the proto.
Bug: chromium:1029452
Change-Id: Iabf841405134d7bc2523ac48219ca7cb9d8214c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170320
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30772}
This reverts commit 30026214b13535a9fe1c47f1463378fbf976c643.
Reason for revert: Speculative revert, breaks downstream test.
Original change's description:
> Fix for out-of-bounds write in square test frame generator.
>
> The length is set on construction and includes an assumption on the
> image resolution, if the resolution changes, a square might be larger
> than what fits into the buffer, causing an out of bounds write. This
> CL fixes this simply by restricting the size of the square.
>
> Bug: webrtc:11415
> Change-Id: Iee14a1971997b4ae2fddef0a7af7c76a2509e879
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170042
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30732}
TBR=srte@webrtc.org,alito@webrtc.org
Change-Id: Ia0056da04a6f6f817ccadfc38aabe0c5f94754cc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11415
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170115
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30743}
The length is set on construction and includes an assumption on the
image resolution, if the resolution changes, a square might be larger
than what fits into the buffer, causing an out of bounds write. This
CL fixes this simply by restricting the size of the square.
Bug: webrtc:11415
Change-Id: Iee14a1971997b4ae2fddef0a7af7c76a2509e879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170042
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30732}
I'll hold on to the root OWNER for a bit longer for convenience.
Bug: None
Change-Id: I13303ba726fed612adc74008eeaaeadf9595e084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170047
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30727}
This is a reland of 49734dc0faa69616a58a1a95c7fc61a4610793cf
Patchset 2 contains a fix for the fuzzer set up. Since we now parse
an RtpPacket out of the fuzzer data, the header needs to be correct,
otherwise we fail before even reaching the FEC code that we actually
want to test.
Bug: webrtc:11340, chromium:1052323, chromium:1055974
TBR=stefan@webrtc.org
Original change's description:
> Reland "Refactors UlpFec and FlexFec to use a common interface."
>
> This is a reland of 11af1d7444fd7438766b7bc52cbd64752d72e32e
>
> Original change's description:
> > Refactors UlpFec and FlexFec to use a common interface.
> >
> > The new VideoFecGenerator is now injected into RtpSenderVideo,
> > and generalizes the usage.
> > This also prepares for being able to genera FEC in the RTP egress
> > module.
> >
> > Bug: webrtc:11340
> > Change-Id: I8aa873129b2fb4131eb3399ee88f6ea2747155a3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168347
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30515}
>
> Bug: webrtc:11340, chromium:1052323
> Change-Id: Id646047365f1c46cca9e6f3e8eefa5151207b4a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168608
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30593}
Bug: webrtc:11340, chromium:1052323
Change-Id: Ib8925f44e2edfcfeadc95c845c3bfc23822604ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169222
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30724}
Replayer isn't triggered in any pre- or post-submit checks
and is built only as a part of fuzzers. Therefore it got out of sync
with the requirement of Call::Config::trials being set.
Bug: chromium:1030755
Change-Id: I467a5fa19137020f6fc748b6adb6f82a8a88f9d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169847
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30695}
The decision to route audio packets to a separate overuse detector
is off by default and requires the field trial
WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/
The parameters control the threshold for switching over to the
audio overuse detector if we stop receiving feedback for video.
Bug: webrtc:10932
Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30694}
