Add peer's name to params and use it for logging and metrics naming
for whole peer related metrics.
Bug: webrtc:11479
Change-Id: Ia7e3fc4839c90a958d66910614515ac02a96e389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174752
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31215}
This CL changes the way that AecDumps are created in APM. Instead
of being injected, they are now created via the API.
This removes the AecDumpFactory from the API surface of APM and
makes the API more explicit.
The CL will be followed by one more CL that deprecates the usage
of the AttachAecDump API also within the audio_processing
and the fuzzer folders.
The CL also moves the aec_dump.* files from the include folder
to the aec_dump folder and changes the build files. The reasons
for this are that
1) The content of aec_dump.h is not really part of the API
surface of APM.
2) Those files anyway needed to be moved to a separate build-
target to avoid a circular build-file dependency caused by
the other changes in this CL
Bug: webrtc:5298
Change-Id: I7dd6b49de76eb44158472874e1d4ae17dca9be54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174750
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31207}
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.
Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
After the migration to passing frame video source implementation directly, part of the peer connection framework code became redundant. Removing screen_share_config and capturing_device_index from the VideoConfig is to be done in later reviews.
Bug: webrtc:11534
Change-Id: I7a8ea85d26d00fb5bfe7ec0d2facef9c44a0f749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174541
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31178}
Implemented an analogue of the cpu_usage metrics from third_party/webrtc/video/video_analyzer.h for third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
Bug: webrtc:11496
Change-Id: Ifdc9daa3351f1df5db98beb8f7dc7156fc7c2a16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174020
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31141}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
Move creation of video sinks for dumping and showing rendered video on
screen into video quality analyzer injection helper to eliminate need
to search for video config in on track callback, which makes this more
reliable and reusable.
Bug: webrtc:11479
Change-Id: I6bb5409688fd39268f9f97bde4c9b0833a64396b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31128}
Extract test peer creation into separate file to simplify code and
increase readability. Also it is 1st step in bigger refactoring of PC
level test fixture implementation to make it more granular and reusable.
Change-Id: I687a17bda33a8eebc1ef0ddc0d54572e095fd709
Bug: webrtc:11479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172628
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30980}
To migrate on new GetStats API and properly support target encode bitrate
for regular, simulcast and svc cases we need to calculate it inside video
quality analyzer getting values from SetRates in VideoEncoder.
Bug: webrtc:11381
Change-Id: Ia37acac764ed3c30f64cdbfda8906d543fa03ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171501
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30881}
Use real video duration instead of test duration to calculate harmonic
frame rate in DefaultVideoQualityAnalyzer.
Bug: webrtc:11445
Change-Id: Ia5f96b2f87178419ec6ebe2ff5dbcb5a0c03c824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171104
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30854}
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}
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}
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}
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}
Introduce FrameGeneratorInterface to make FrameGenerator API available
for downstream projects.
Bug: webrtc:10138
Change-Id: I4216775e4b8b54c3f1c72d67ffbda31eb082fd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161234
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30009}
Add ability to provide custom implementation of rtc::VideoSourceInterface
as source for video track in PC-framework based media quality tests.
Bug: webrtc:10138
Change-Id: I8ffd3015230c733a0a9a2e97fd4bb93a0c02b283
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159680
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29776}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
