The IDs be more stable than the plot titles and could be used to identify specific graphs in scripts.
Change event_log_visualizer command line interface to control which plots are generated.
Old interface had one command line flag per plot as well as a set of 'profiles' that enabled
of disabled sets of plots. New interface has a command line flag
which takes a string of all the plot names or profiles that should be enabled.
In some cases, there are also slight naming changes for the plots.
For example, the former command
event_log_visualizer --plot_profile=sendside_bwe --plot_incoming_packet_sizes <filename> | python
is now
event_log_visualizer --plot=sendside_bwe,incoming_packet_sizes <filename> | python
The former command
event_log_visualizer --plot_profile=none --plot_incoming_packet_sizes <filename> | python
is now
event_log_visualizer --plot=incoming_packet_sizes <filename> | python
The former command
event_log_visualizer --plot_profile=all <filename> | python
is now
event_log_visualizer --plot=all <filename> | python
Bug: webrtc:10623
Change-Id: Ife432c1e51edfce64af565a769f1764a16655bb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140886
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28237}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
When running unpack_aecdump --full, unpack RuntimeSettings into files, on the format that can be imported into Audacity.
Output one file for each RuntimeSetting present in the aecdump. If outputting several WAV files, output file for each WAV file with corresponding time stamps.
Bug: webrtc:10643
Change-Id: If147e509d36207f5f838457354e2451df65549d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137426
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28007}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This CL adds a mode to simulate roughly what GoogCC could have been
doing during the recording of an rtc event log by using the logged
events as input to GoogCC and visualizing the resulting target rate.
This is similar to the existing simulated_sendside_bwe mode, but uses
the new NetworkControllerInterface to ensure more reliable GoogCC
simulation.
Bug: None
Change-Id: I57894aa666151efc8405407d928b5257fb9b7d61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123924
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27095}
Import proto_library.gni when rtc_enable_protobuf is true instead of when
build_with_mozilla is false.
Makes it maybe easier to reason about the intention (e.g. intention is to not
compile any protobuf in, hence flag rtc_enable_protobuf)
The build file could not work if build_with_mozilla = true but
rtc_enable_protobuf = true.
Bug: webrtc:10338
Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597
Reviewed-on: https://webrtc-review.googlesource.com/c/123763
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26868}
This CL introduces a new rtp_generator tool that can be utilized to generate
.rtpdump files that can be replayed by the video_replayer. This allows
automated generation of corpus material for the new WebRTC RTP fuzzers in
addition to allowing anyone who is experimenting with a new RTP feature to
quickly debug issues.
It can be used as follows:
./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump
./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump
It works by generating squares randomly on the screen for a given duration. This
initial version is very limited and doesn't support FEC, RED and other
configurations. I plan to extend it to support these in future CLs.
Bug: webrtc:10117
Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51
Reviewed-on: https://webrtc-review.googlesource.com/c/119964
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26517}
The WebRtcKeyValueConfig interface allows providing custom key value
configurations that changes per instance of GoogCcNetworkController.
Bug: webrtc:10009
Change-Id: I520fff030d1c3c755455ec8f67896fe8a6b4d970
Reviewed-on: https://webrtc-review.googlesource.com/c/116989
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26312}
This class adds logic for aligning what part of a test video has been
encoded from a reference video. It does that by cropping and zooming in
on a region of the reference video that most closely matches the test
video. A small cropping does not have much impact on human perception,
but it has a big impact on PSNR and SSIM calculations.
For example, if the test video is cropped with one row in the top and
bottom, adjusting for this improves average PSNR from 27.7146 to
29.3357 and average SSIM from 0.934891 to 0.95318 in an example test
video.
TBR=phoglund
Bug: webrtc:9642
Change-Id: I02cfe0e2261fb58df8cdb1e15ba93285e3dc4538
Reviewed-on: https://webrtc-review.googlesource.com/c/99480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25755}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
This class adds logic for aligning color space of a test video compared
to a reference video. If there is a color space mismatch, it typically
does not have much impact on human perception, but it has a big impact
on PSNR and SSIM calculations. For example, aligning a test run with VP8
improves PSNR and SSIM from:
Average PSNR: 29.142818, average SSIM: 0.946026
to:
Average PSNR: 38.146229, average SSIM: 0.965388.
The optiomal color transformation between the two videos were:
0.86 0.01 0.00 14.37
0.00 0.88 0.00 15.32
0.00 0.00 0.88 15.74
which is converting YUV full range to YUV limited range. There is
already a CL out for fixing this discrepancy here:
https://webrtc-review.googlesource.com/c/src/+/94543
After that, hopefully there is no color space mismatch when saving the
raw YUV values. It's good that the video quality tool is color space
agnostic anyway, and can compensate for differences when the test
video is obtained by e.g. filming a physical device screen.
Also, the linear least square logic will be used for compensating
geometric distorisions in a follow-up CL.
Bug: webrtc:9642
Change-Id: I499713960a0544d8e45c5d09886e68ec829b28a7
Reviewed-on: https://webrtc-review.googlesource.com/c/95950
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25193}
Extract functionality of test_main into separate library to be able to
reuse it if another main will be required.
Bug: webrtc:5996
Change-Id: I2925b4240bd0e4fb884b43bb16667ca2d6216bbd
Reviewed-on: https://webrtc-review.googlesource.com/c/105921
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25172}
This tool is no longer needed since we're deleting the AQ tests.
Bug: chromium:880074
Change-Id: I035d7b33c7c4feb5962cf9dafc8e7086a8dee440
Reviewed-on: https://webrtc-review.googlesource.com/c/105140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25162}
And also drop dependency on module_api, where possible. With this
change, common_video/ no longer depends on
libjingle_peerconnection_api.
Bug: None
Change-Id: Icc0648559bef5b7f549e81d58f2a5f97c0af3abf
Reviewed-on: https://webrtc-review.googlesource.com/c/103782
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24991}
After the removal of field_trial_default, metrics_default and
runtime_enabled_features_default, this build target doesn't build
anything and can be safely removed.
Bug: webrtc:9631
Change-Id: Iee1111e065ffefe0b4b9a695ee67a594e6d82caa
Reviewed-on: https://webrtc-review.googlesource.com/c/103702
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24976}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
The function checks the file extension to determine YUV or Y4M format.
Also adds a flag aligned_output_file to compare_videos.py, which allows
saving the aligned reference video to a file.
Bug: webrtc:9642
Change-Id: Ia59f5c123a1e41104756eb6b235b6581c4ffbd77
Reviewed-on: https://webrtc-review.googlesource.com/99503
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24787}
This is a reland of 9bb55fc09b6bfa00cba7779c37ad6c39b4206f7a
Original change's description:
> Reland "Update video_quality_analysis to align videos instead of using barcodes"
>
> This is a reland of d65e143801a7aaa9affdb939ea836aec1955cdcc
>
> The binary for frame_analyzer.cpp is precompiled and stored in the cloud, so it
> won't automatically pick up change to the source file. Therefore, restore all
> old code to be backwards compatible.
>
> Original change's description:
> > Update video_quality_analysis to align videos instead of using barcodes
> >
> > This CL is a follow-up to the previous CL
> > https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> > logic for aligning videos. This will allow us to easily extend
> > video_quality_analysis with new sophisticated video quality metrics.
> > Also, we can use any kind of video that does not necessarily need to
> > contain bar codes. Removing the need to decode barcodes also leads to a
> > big speedup for the tests.
> >
> > Bug: webrtc:9642
> > Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> > Reviewed-on: https://webrtc-review.googlesource.com/94845
> > Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24423}
>
> TBR=phensman@webrtc.org,phoglund@webrtc.org
>
> Bug: webrtc:9642
> Change-Id: Id8d129ce103284504c67690f8363c03eaae3eee7
> Reviewed-on: https://webrtc-review.googlesource.com/96000
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24429}
TBR=phensman,phoglund
Bug: webrtc:9642
Change-Id: Ic248b7831ae148251a1a4ebeec5d154286f91a0a
Reviewed-on: https://webrtc-review.googlesource.com/98080
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24583}
