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}
This is to satisfy a thread checker in AudioSendStream.
Bug: webrtc:9510
Change-Id: I5ba03562fcdc3e93d77707e41220b82b99581470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169343
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30648}
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 go from 7 seconds each to 2 seconds each with this change, and
I belive they will catch correctness bugs just as well.
With this and https://webrtc-review.googlesource.com/c/src/+/168884,
test_support_unittests now runs in 14 seconds instead of 65 (in
sequential mode).
Bug: None
Change-Id: Ic04e3937bbff54f33dcd062f422dada176f1c3cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168885
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30577}
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}
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}
The EncoderSelectorInterface is meant to replace the "WebRTC-NetworkCondition-EncoderSwitch" field trial, so the field trial will be ignored if an EncoderSelectorInterface object has been injected.
Bug: webrtc:11341
Change-Id: I5371fac9c9ad8e38223a81dd1e7bfefb2bb458cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168193
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30490}
This is a reland of 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2
Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
>
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
>
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
>
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
>
> This allows containing the logic fully within RTPSenderVideo.
>
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}
TBR=stefan@webrtc.org
Bug: webrtc:11340
Change-Id: I2fdd0004121b13b96497b21e052359e31d0c477a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168305
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30479}
This reverts commit 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2.
Reason for revert: Breaks downstream project
Original change's description:
> Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery
>
> The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
> header extension was successfully propagated to the receiving side. Once
> it was determined that the receiver had received a frame with the new
> delay tag, it's no longer necessary to propagate.
>
> The issue with this implementation is that it is based on max
> extended sequence number reported via RTCP, which makes it often slow
> to react, could theoretically fail to produce desired outcome (max
> received > X does not guarantee X was fully received and decoded), and
> added a lot of code complexity.
>
> The guarantee of delivery can in fact be accomplished more reliably and
> with less code by making sure to tag each frame until an undiscardable
> frame is sent.
>
> This allows containing the logic fully within RTPSenderVideo.
>
> Bug: webrtc:11340
> Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30473}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ide922e680ae36bb69b95e58002482cf5ed57e254
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168304
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30475}
The PlayoutDelayOracle was responsible for making sure the PlayoutDelay
header extension was successfully propagated to the receiving side. Once
it was determined that the receiver had received a frame with the new
delay tag, it's no longer necessary to propagate.
The issue with this implementation is that it is based on max
extended sequence number reported via RTCP, which makes it often slow
to react, could theoretically fail to produce desired outcome (max
received > X does not guarantee X was fully received and decoded), and
added a lot of code complexity.
The guarantee of delivery can in fact be accomplished more reliably and
with less code by making sure to tag each frame until an undiscardable
frame is sent.
This allows containing the logic fully within RTPSenderVideo.
Bug: webrtc:11340
Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30473}
This will be used by WebRTC tests. It converts results exactly the
same as our downstream implementation (histogram_util).
This implementation should be pretty feature complete, or at least
enough to start testing the end-to-end flow. I will set up some
experimental recipe code and see if this actually makes it into the
dashboard.
Note: needs some catapult changes to land first and be rolled
into Chromium, and then WebRTC.
Bug: chromium:1029452
Change-Id: I939046929652fc27b8fcb18af54bde22886d9228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166172
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30436}
A previous refactoring introduced an issues in SimulatedProcessThread
causing stalls when task are posted. This CL fixes this and cleans up
the code to make it easier to see that it's correct.
Bug: webrtc:11255
Change-Id: I33d7daa993ad2a4cfe2b63f674692455c2e09d05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167380
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30429}
Also updated FrameBuffer unittests to use the GlobalSimulatedTimeController.
Bug: webrtc:7408, webrtc:9378
Change-Id: I8ade27492f66cdd8950b38f5f4a268714dbc35fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164536
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30422}
I changed stuff in test/BUILD.gn, but the suggested formatting broke
the presubmit. I tried rewriting the presubmit so it checks the
previous line as well, but that turned out to be hard.
Please try to enable this presubmit on ALL lines in a changed file.
Presubmits that only work on changed lines are really confusing.
Bug: None
Change-Id: I2386c765367681f683d82739293bc8bc8a873a7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167926
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30420}
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.
Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
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@{#30404}
