Old target and call/simulated.h exist but refer to new target in test/network.
Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
This adds Perfetto support to WebRTC with a GN flag rtc_use_perfetto.
The configuration of perfetto depends on whether or not webrtc is
build within Chrome or not. When in Chrome, WebRTC will depend on
//third_party/perfetto:libperfetto. When building standalone, specific includes required for Perfetto are exposed with the library webrtc_libperfetto.
The perfetto trace API is exposed with a header export in
trace_event.h which is used instead of the legacy API.
The addition of Perfetto means there are 4 compilation modes for
tracing in WebRTC,
1. No tracing implementation.
2. Legacy tracing (AddTraceEvent/GetCategoryEnabled).
3.a. Perfetto statically linked (webrtc_libperfetto).
3.b. Perfetto in Chrome (Chrome's libperfetto).
This CL removes the tracing expectations from
rtc_stats_integrationtest.cc because those directly used the old API.
Integration into Chrome is a follow up CL which depends on
https://chromium-review.googlesource.com/c/chromium/src/+/5471691.
Tested: Ran Chrome with Perfetto and traces appear. WebRTC Unit test tracing working: https://ui.perfetto.dev/#!?s=04ea2613ea36b814394639a1ec4b60be5b5097527f1a485995ecc13469885468
Bug: webrtc:15917
Change-Id: I537d79dc247c2b759689910c621087286a4d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mikhail Khokhlov <khokhlov@google.com>
Cr-Commit-Position: refs/heads/main@{#42166}
Some of the fake encoders, FakeVp8Encoder in particular, reuse structures that in turn rely on field trials. Thus fake encoders also can benefit from Environment passed at construction.
Bug: webrtc:15860
Change-Id: Ia1542b2663c75fd467e346aad9ead627ff9b3b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346780
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42046}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
This approach actually wraps the unique identifier generation into the
function that provides the output path for a test.
This way we don't need to add `CreateRandomUuid()` everywhere that we
have `test::OutputPath` and instead just rename to
`test::OutputPathRandomDir`
Bug: webrtc:15833
Change-Id: Ic9b69b5b599727f07b2906569a84a40edeecd1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41730}
Providing unique identifiers for files and directories created as part
of unit tests.
Bug: webrtc:15833
Change-Id: If2835c362c47a111aa99b0e3c6ad6a33be061978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41704}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
Create one decoder per simulcast stream and pass encoded frame to a dedicated decoder.
Bug: webrtc:14852
Change-Id: I2a0baaa1e28b38507993eb4269b15ae89695d670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41439}
* Add VideoCodecTesterTest and move Run[EncodeDecode]Test() into it. The class will be extended with functionality necessary for testing simulcast/SVC (it will collect and store decode input frame sizes in particular) in follow-up CLs, which will add simulcast/SVC support to the tester.
* Add TestVideoEncoder and TestVideoDecoder classes.
* Use frame size instead of timestamp in checks in Slice test. Unlike timestamp, which has the same value for spatial layer frames within a temporal unit, frame size is a unique frame property in these tests.
Bug: webrtc:14852
Change-Id: I2386183688dd4988ca56e0ab53edbb9f5fcf6c9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331362
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41438}
This is a reland of commit 496893e89e5bc8139e50befcb1a26eadbd829b0d
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Iccb9af8bf6a6c37704bc58b6e57238b55761b079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41194}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
This field trial is configured via command line flag, so may use flag system directly, reducing dependency on global field trial string.
Bug: webrtc:7101, webrtc:10335
Change-Id: I1e48e0e3fdc251b73a375c6d7f1a46fa4f8a179b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322624
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40897}
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.
The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.
Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262
WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.
This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.
Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
The following can lead to ODR violations with symbols present in the
app and in the test module:
gn path out/Perf //:webrtc_perf_tests_module //sdk:helpers_objc
//:webrtc_perf_tests_module --[public]-->
//:webrtc_perf_tests_module_loadable_module --[private]-->
//test:google_test_runner_objc --[private]-->
//test:test_support_objc --[private]-->
//sdk:helpers_objc
After this CL:
gn path out/Debug/ //:webrtc_perf_tests_module //sdk:helpers_objc
No non-data paths found between these two targets.
Bug: b/292472934
Change-Id: If8a6ecab9b34bea0f52fe91b3404d1afeca685fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313520
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40481}
It is partial reland, which adds call to Start() to all relevant places,
but doesn't actually switches frame generator to not produce frames from
the moment it was created.
Bug: b/272350185
Change-Id: I6e3bd7af6f5cd8d9baff79c2aada7b2ddfae1c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310782
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40379}
This CL implements {,Logging}DelayVariationCalculator, whose purpose is to calculate simple inter-arrival metrics for a sequence of RTP frames. Uses could include RtcEventLog analysis and ad hoc testing.
Want lgtm: asapersson
Bug: webrtc:15213
Change-Id: I3f9d13a2c4fa66b6f1229c1b6fcd66a6911070de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306741
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40247}
This is step to allow migration of Test ADM to the AudioDeviceModuleImpl
as a base class to include AudioDeviceBuffer into SUT.
Also it will allow to remove WaitForRecordingEnd() method from Test
ADM
Bug: b/272350185, webrtc:15081
Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39849}
Add a WebRTC-specific arg that can be used to control use of targets
that rely on //third_party/google_benchmarks, so the .gni in that
directory can eventually be removed.
Bug: chromium:1404759
Change-Id: I2a9422fae119ca13eb50028d962fc0a671b5fb33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39553}
