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}
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}
Making these tests run shorter broke them on iOS. I think we can just
be more tolerant on iOS.
This also tried to re-enable the test on dbg; hopefully the increased
tolerance is enough.
Bug: None
Change-Id: Ic8c54dd46b0f5cb219b0c16da81c9486f6c45f10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169440
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30660}
Extract collection of BWE stats from DefaultVideoQualityAnalyzer to
separate class to prepare for migration on new GetStats API and simplify
quality analyzer.
Bug: webrtc:11381
Change-Id: I0e7e2d7e40b467d7a42633a72a7ffc49ebcb0237
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169444
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30650}
One should use a std::unique_ptr to the object, as returned
by Clone() instead, not a naked pointer.
Bug: webrtc:10701
Change-Id: I10ab309207f2cb5aec83a6d09336699ed7b26f50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169342
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30646}
Move FuzzAudioEncoder to a separate target to make it available for
other encoders.
Bug: webrtc:11388
Change-Id: I8b9a0f810791880eedb129b55eb33f154790e48f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169364
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30634}
This makes it easier to reuse existing audio network adaptation
configurations in the scenario framework.
Bug: webrtc:9510
Change-Id: I06ab08684d449fef7fffe265d1078738d526a43d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169363
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30633}
This makes slightly more sense when looking at video resolution etc.
Bug: webrtc:9510
Change-Id: I49d39cac23d2f5d7ca09f2a27152c7519ea639f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169344
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30632}
That would allow to switch components from relying on ProcessThreads to
relying on TaskQueue one by one, without introducing new threads.
Bug: webrtc:6289
Change-Id: I18fe5d679d4d4d0ddf4a11900c9814eb570284d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167533
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30631}
This reverts commit 49734dc0faa69616a58a1a95c7fc61a4610793cf.
Reason for revert: Still something wrong with ulpfec fuzzer setup.
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}
TBR=sprang@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11340, chromium:1052323
Change-Id: I920ce0a48a08768d7a98a563e2b66bd6eb8602b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169121
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30616}
frame_id is unwraped from a 16bit value.
Getting to int64_t boundaries would take more than 2^48 packets.
That scenario considered unrealistic and thus untested.
Bug: chromium:1053482
Change-Id: Ib3f52d4528b20915b2330773f616d9304f45cac6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168682
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30607}
DefaultVideoQualityAnalyzer accumulates in flight frames in internal
queue to perform psnr/ssim computation. This queue can grow a lot if
test experience high frame loss. As a result of this, the analyzer
can use quite a lot of memory and cause OOM crashes.
This CL limits the size of the queue based on the assumption that after
a certain point a frame can be considered lost and so it is impossible
to calculate PSNR/SSIM.
Bug: webrtc:11373
Change-Id: Iaabcc8d1c3c9142dc58ea5f2f30f599864b088e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168951
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30602}
This is the WebRTC equivalent of testing/perf/perf_result_reporter.h
in Chromium. That class was introduced because the PrintResult
functions are quite hard to use right. It was easy to mix up
metrics, modifiers and stories, for instance.
I choose to introduce this new class because I need to create a new
API for PrintResult anyway. For instance, the important bool isn't
really supported by histograms. Also I would like to restrict units
to an enum because you cannot make up your own units anymore.
We could also have had a strictly checked string type, but that's
bad API design. An enum is better because the compiler will check
that the unit is valid rather than at runtime.
Furthermore, down the line we can probably make each reporter write
protos directly to /tmp and merge them later, instead of having a
singleton which writes results at the end and keeps all test results
in memory. This abstraction makes it easy to make a clean and simple
implementation of just that.
Steps:
1) land this
2) start rewriting perf tests to use this class
3) nuke PrintResult functions
4) don't convert units to string, convert directly from Unit
to proto::Unit
5) write protos directly from this class (either through
a singleton or directly) and nuke the perf results writer
abstraction.
Bug: chromium:1029452
Change-Id: Ia919c371a69309130c797fdf01ae5bd64345ab2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168770
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30599}
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}
I think these functions are so hard to understand, so I tried to
make an as grounded example as possible.
Bug: chromium:1029452
Change-Id: I5d4284bc15b39cb94ba42d2c483a619ecf42fb91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168945
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30590}
They now run in 3 seconds rather than 45 or whatever it was before.
The tests still pass (and I tried with gtest_repeat=25), so I think
the shorter time is sufficient to prove the code works and doesn't
crash. Unit tests need to be fast. I think it's unlikely a longer
runtime would make this test a better correctness test, but let me
know if there's something in particular with this code that needs
the longer runtime.
Bug: None
Change-Id: I3f4213718870a1772f7a19e3c418634031c46de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168884
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30569}
Turns out that MacOS 10.14.6 requires CFBundleShortVersionString (it
refuses to install the app if the string isn't there).
This should fix the iOS 64-bit bots.
Bug: chromium:1053891
Change-Id: I3278502eff9813fed9a2d8e442c940dfb70377cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168882
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30567}
Force copy video frame including video buffer in
DefaultVideoQualityAnalyzer to ensure that analyzer won't hold any
internal WebRTC buffers.
Bug: webrtc:10138
Change-Id: Ib195233f8b01c855220be1b9743c4f54fc62a22b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168643
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30535}
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.
This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
with adjustments.
Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
This reverts commit 11af1d7444fd7438766b7bc52cbd64752d72e32e.
Reason for revert: Possible crash
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}
TBR=brandtr@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Iddf112d801621c8a4370b853cee3fa42bf2c7fba
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30524}
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}
This reverts commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
Reason for revert: breaks downstream tests
Original change's description:
> Do not propagate generic descriptor on receiving frame
>
> It was used only for the frame decryptor.
> Decryptor needs only raw representation that it can recreate
> in a way compatible with the new version of the descriptor.
>
> Bug: webrtc:10342
> Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30501}
TBR=danilchap@webrtc.org,sprang@webrtc.org,philipel@webrtc.org
Change-Id: I6634df06ee75aa8cdfda614994ab11f7a5845c70
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168488
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30502}
