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}
Since RTC_DCHECK was made constexpr compatible, we can now
make the unit classes fully constexpr.
Bug: webrtc:9883
Change-Id: I18973c2f318449869cf0bd45699c41be53fba806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30403}
The --trace_event=file.json option allows to log events,
for further inspection in chromium event viewer.
Previous handling of this option was broken,
closing the logger before the tests were even run.
Bug: webrtc:10926
Change-Id: I9123d12666b5f254feeaef685def96eb8ba1c7f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30401}
This makes it easier to maintain consistency between real time
and simulated time modes.
The RealTimeController is updated to use an explicit main thread,
this ensures that pending destruction tasks are run as the network
emulator goes out of scope.
Bug: webrtc:11255
Change-Id: Ie73ab778c78a68d7c58c0f857f14a8d8ac027c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30342}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This is a reland of 2a11b2451a4068746fa0c55fa210efd4a15e4423
There are no changes compared to the first attempt.
Original change's description:
> Enable using a custom NetEqFactory in simulations
>
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}
TBR=kwiberg
Bug: webrtc:11005
Change-Id: I4aa377e05916bd23f8f63aece9d0e27731c80d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166465
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30319}
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.
Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
This is a reland of f3aa6326b8e21f627b9fba72040122723251999b
Original change's description:
> Replace the ExperimentalAgc config with the new config format
>
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
>
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
>
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}
Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
This is a reland of b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453
Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
>
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}
Bug: webrtc:11255
Change-Id: If65cd56b59158cebec5609407a721fbdb47cfd1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30294}
