This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
- To achieve this, WebRTC encoders are updated to map kNative video
buffers so that in a follow-up CL VideoStreamEncoder can stop mapping
intermediate buffer sizes.
In this CL LibvpxVp9Encoder is updated to map kNative buffers of pixel
formats it supports and convert ToI420() if the kNative buffer is
something else. A fake native buffer that keeps track of which
resolutions were mapped, MappableNativeBuffer, is added.
Because VP9 is currently an SVC encoder and not a simulcast encoder, it
does not need to invoke CropAndScale.
This CL also fixes MultiplexEncoderAdapter, but because it simply
forwards frames it only cares about the pixel format when
|supports_augmented_data_| is true so this is the only time we map it.
Because this encoder is not used with kNative in practise, we don't care
to make this path optimal.
Bug: webrtc:12469, chromium:1157072
Change-Id: I74edf85b18eccd0d250776bbade7a6444478efce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212580
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#33526}
This avoids falling back on the VideoFrameBuffer::CropAndScale default
implementation which performs ToI420. This has two major benefits:
1. We save CPU by not converting to I420 for NV12 frames.
2. We make is possible for simulcast encoders to use Scale() and be
able to trust that the scaled simulcast layers have the same pixel
format as the top layer, which is required by libvpx.
In order to invoke NV12Buffer::CropAndScaleFrom() without introducing a
circular dependency, nv12_buffer.[h/cc] is moved to the "video_frame"
build target.
Bug: webrtc:12595, webrtc:12469
Change-Id: I81aac5c6b3e81c49f32a7be6dc2640e6b40f7692
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33521}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
Some targets depends on targets under enable_google_benchmarks. But they
are not under such if statement themeself.
Bug: webrtc:12404
Change-Id: I7c0b9a75bd3fa18090ef6a44fda22ed5f33d79b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204063
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33104}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
(Reland with no changes after the fix to the downstream project)
This can be overriden for kNative frame types to perform scaling efficiently.
Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.
Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org
Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
Currently isolated output directory is created in flags_compatibility.py script.
This doesn't work for android swarming tasks because this script isn't called.
Bug: webrtc:11895
Change-Id: I8b8f01850d6e5970292b524d104314eef7ab17be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185883
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32236}
This can be used in the future to test NV12 video frames with encoders, both
from unittests and from tools like video_loopback.
Tested using video_loopback with generator NV12.
Bug: webrtc:11978
Change-Id: I0d24ae3ebab2267f076703cbda81e99cec465ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185045
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32206}
I ran into this when using repeating_task, which depends on clock (in
system_wrappers) which in turn added a dependency on rtc_base on Windows
due to win32 files. That's a problem since rtc_base depends on
repeating_task:
//rtc_base:rtc_base ->
//rtc_base/task_utils:repeating_task ->
//system_wrappers:system_wrappers ->
//rtc_base:rtc_base
We could additionally consider moving Clock out of system_wrappers.
Bug: webrtc:9987
Change-Id: I54ed715ad5eb9e3f5dd6c322233c18c05d895dff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185506
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32203}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
GN recently added support for Apple frameworks to link, rather than
overloading the libs lists. This pulls .frameworks out of the libs
lists, so that GN can stop supporting .frameworks in libs in the
future.
Bug: chromium:1052560
Change-Id: I263230ddd3c468061584423bba9e1f887503bcaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178601
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/master@{#31632}
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.
The mutex types supportable by webrtc::Mutex are
- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)
In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.
The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.
Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.
Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
Also re-enable the TestAnnotationsOnWrongQueueDebug test and rename
the test suite to SequenceCheckerDeathTest so that it gets executed
before other tests.
Bug: webrtc:11577
Change-Id: I3b8037644e4b9139755ccecb17e42b09327e4996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175346
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31290}
on the current thread.
This simplifies writing async tests that use TaskQueue and doesn't
require spinning up a new thread for simple things. The implementation
is currently based on rtc::Thread, which could also be useful in
some circumstances while migrating code over to TQ.
Remove PressEnterToContinue from the test_common files since
it's very specific and only used from one file.
Bug: none
Change-Id: I8b2c6c40809271a109ec17cf7e1120847645d58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31160}
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}
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}
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 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}
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}
`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}
The new interface is called PerfTestResultWriter and is currently
implemented by PerfResultsLogger (renamed PerfTestGraphJsonWriter).
I plan to introduce a second implementation of the perf logger that
uses the new Histogram C++ API. I add a flag that chooses
between the two implementations so I can try it out (perhaps by
setting up a second, limited run of webrtc_perf_tests on the perf
bots that uses the new implementation). The histogram C++
implementation will come in the next patch.
As a side effect, I disentangled the plottable counter printer from
the perf result printer so it will work for both implementations.
The only thing they had in common was that both wrote JSON anyway.
See the bug for details on the new API.
Bug: chromium:1029452
Change-Id: Icb21b25ced08ea73aeecd221e9d51f2adf3dab1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165389
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30243}
Replace all usages of java_files with sources in gn files, and
automatically format.
This is in preparation for java_files being completely removed upstream
in favor of sources.
NOPRESUBMIT=true
Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.
Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
