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webrtc_m130/rtc_tools/BUILD.gn

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Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../webrtc.gni")
Import proto_library.gni when rtc_enable_protobuf is true 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}
2019-02-26 11:46:17 +02:00
if (rtc_enable_protobuf) {
import("//third_party/protobuf/proto_library.gni")
}
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
group("rtc_tools") {
# This target shall build all targets in tools/.
testonly = true
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
deps = [
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
":frame_analyzer",
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
":video_file_reader",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
if (!build_with_chromium) {
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
deps += [
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
":psnr_ssim_analyzer",
":rgba_to_i420_converter",
Switch to absl single target when building with Chromium. The //third_party/abseil-cpp:absl target is currently a group that depends on all the targets needed by WebRTC in Chromium. It will be switched to a component starting from https://chromium-review.googlesource.com/c/chromium/src/+/2174434. Bug: chromium:1046390 Change-Id: I70d450fdbfa895084b481c9884b6361d2fb9580d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176901 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31498}
2020-06-11 11:25:32 +02:00
":video_quality_analysis",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
]
if (rtc_enable_protobuf) {
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
deps += [ ":chart_proto" ]
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
}
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
}
if (rtc_include_tests) {
Add YUV to IVF video converter util Bug: webrtc:10138 Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 15:03:45 +01:00
deps += [
":tools_unittests",
":yuv_to_ivf_converter",
]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
if (rtc_enable_protobuf) {
Harmonize paths to dependent targets. 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}
2018-10-30 21:12:42 +01:00
if (!build_with_chromium) {
deps += [ ":event_log_visualizer" ]
}
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
deps += [
Move audioproc_f to rtc_tools. The motivation in https://webrtc-review.googlesource.com/c/src/+/32340/3 applies here as well. We would like to use this tool downstream. Bug: None Change-Id: Id5b23f792679ab9c07294bfb8e53119c423044b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161681 Commit-Queue: Daniel Johansson <dajo@webrtc.org> Reviewed-by: Daniel Johansson <dajo@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30051}
2019-12-10 11:00:38 +01:00
":audioproc_f",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
":rtp_analyzer",
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
":unpack_aecdump",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"network_tester",
]
}
}
}
Use source_sets in component builds and static_library in release builds. 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}
2019-10-17 21:32:04 +02:00
rtc_library("video_file_reader") {
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
sources = [
"video_file_reader.cc",
"video_file_reader.h",
]
deps = [
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. 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}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../rtc_base:checks",
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
"../rtc_base:rtc_base_approved",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Replace starts_with and ends_with with Abseil Bug: None Change-Id: I7eae3db1aeb81f0f1d37ff50d5c85c16ecb1f366 Reviewed-on: https://webrtc-review.googlesource.com/c/114221 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26032}
2018-12-13 17:41:25 -08:00
"//third_party/abseil-cpp/absl/strings",
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
"//third_party/abseil-cpp/absl/types:optional",
]
}
Use source_sets in component builds and static_library in release builds. 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}
2019-10-17 21:32:04 +02:00
rtc_library("video_file_writer") {
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
sources = [
"video_file_writer.cc",
"video_file_writer.h",
]
deps = [
":video_file_reader",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. 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}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
"../rtc_base:rtc_base_approved",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Replace starts_with and ends_with with Abseil Bug: None Change-Id: I7eae3db1aeb81f0f1d37ff50d5c85c16ecb1f366 Reviewed-on: https://webrtc-review.googlesource.com/c/114221 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26032}
2018-12-13 17:41:25 -08:00
"//third_party/abseil-cpp/absl/strings",
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
"//third_party/abseil-cpp/absl/types:optional",
]
}
Use source_sets in component builds and static_library in release builds. 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}
2019-10-17 21:32:04 +02:00
rtc_library("video_quality_analysis") {
Extract an interface from the perf results logger. 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}
2020-01-13 15:10:40 +01:00
testonly = true
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
sources = [
Add tool for aligning color space of video files 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}
2018-09-11 12:56:06 +02:00
"frame_analyzer/linear_least_squares.cc",
"frame_analyzer/linear_least_squares.h",
"frame_analyzer/video_color_aligner.cc",
"frame_analyzer/video_color_aligner.h",
Add tool for aligning cropped region of video files 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}
2018-11-22 15:17:30 +01:00
"frame_analyzer/video_geometry_aligner.cc",
"frame_analyzer/video_geometry_aligner.h",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"frame_analyzer/video_quality_analysis.cc",
"frame_analyzer/video_quality_analysis.h",
Reland "Add tool for aliging video files" This is a reland of b2c0e8f60fad10e2786e5e131136a0da1299d883 Original change's description: > Add tool for aliging video files > > This class adds logic for aligning a test video to a reference video > by an algorithm that maximizes SSIM between them. Aligned videos will be > easier to run video quality metrics on. This is a generic way of > aligning videos and can be replace the intrusive barcode stamping that > we currently use. This will be done in a follow-up CL. > > Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94773 > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24407} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: I35d6b0e598335b8d80fbfa37ba06d5c651bda4f6 Reviewed-on: https://webrtc-review.googlesource.com/98040 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24580}
2018-09-05 15:04:08 +02:00
"frame_analyzer/video_temporal_aligner.cc",
"frame_analyzer/video_temporal_aligner.h",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
Stop using public_deps to depend on libyuv. A lot of WebRTC targets were depending on //third_party/libyuv using public_deps instead of deps. This causes issues because a the inclusion of libyuv headers is not declared to the build system and this creates hidden dependencies that put the modularity of the project at risk. Bug: webrtc:8603 Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/28120 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21039}
2017-12-01 13:53:04 +01:00
deps = [
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
":video_file_reader",
Add tool for aligning color space of video files 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}
2018-09-11 12:56:06 +02:00
"../api:array_view",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Add tool for aligning cropped region of video files 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}
2018-11-22 15:17:30 +01:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. 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}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
"../common_video",
Reland "Add tool for aliging video files" This is a reland of b2c0e8f60fad10e2786e5e131136a0da1299d883 Original change's description: > Add tool for aliging video files > > This class adds logic for aligning a test video to a reference video > by an algorithm that maximizes SSIM between them. Aligned videos will be > easier to run video quality metrics on. This is a generic way of > aligning videos and can be replace the intrusive barcode stamping that > we currently use. This will be done in a follow-up CL. > > Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94773 > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24407} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: I35d6b0e598335b8d80fbfa37ba06d5c651bda4f6 Reviewed-on: https://webrtc-review.googlesource.com/98040 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24580}
2018-09-05 15:04:08 +02:00
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
Store video_quality_loopback_test perf results in Chart JSON format. Adds a flag to store the perf results in a JSON file using the Chart JSON format [1]. [1] https://github.com/catapult-project/catapult/blob/master/dashboard/docs/data-format.md TBR=phoglund@webrtc.org Bug: chromium:755660 Change-Id: I6a896654a4a558df217ddefa4e8a52a487cdbebd Reviewed-on: https://webrtc-review.googlesource.com/43180 Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21809}
2018-01-30 15:33:02 +01:00
"../test:perf_test",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
}
Add absl_deps field for rtc_test and rtc_executable To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl". Bug: webrtc:12404 Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 10:50:19 +00:00
# Abseil dependencies are not moved to the absl_deps field deliberately.
# If build_with_chromium is true, the absl_deps replaces the dependencies with
# the "//third_party/abseil-cpp:absl" target. Which doesn't include absl/flags
# (and some others) because they cannot be used in Chromiums. Special exception
# for the "frame_analyzer" target in "third_party/abseil-cpp/absl.gni" allows
# it to be build in chromium.
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
rtc_executable("frame_analyzer") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Reland "Switch frame_analyzer to ABSL_FLAG." This is a reland of 3668e9b56198eb8112562d017c47a52050be3561 Original change's description: > Switch frame_analyzer to ABSL_FLAG. > > Bug: webrtc:10616 > Change-Id: I34473632431569406b9086f57caac12434862005 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136801 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28358} TBR=kwiberg@webrtc.org Bug: webrtc:10616 Change-Id: I661ec82938c80964382fcd0692cf9d5cdb132977 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143681 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28361}
2019-06-18 21:09:24 +02:00
testonly = true
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
sources = [ "frame_analyzer/frame_analyzer.cc" ]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
deps = [
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
":video_file_reader",
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
":video_file_writer",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
":video_quality_analysis",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Reland "Reland "Update video_quality_analysis to align videos instead of using barcodes"" 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}
2018-09-05 16:11:48 +02:00
"../rtc_base:stringutils",
Store video_quality_loopback_test perf results in Chart JSON format. Adds a flag to store the perf results in a JSON file using the Chart JSON format [1]. [1] https://github.com/catapult-project/catapult/blob/master/dashboard/docs/data-format.md TBR=phoglund@webrtc.org Bug: chromium:755660 Change-Id: I6a896654a4a558df217ddefa4e8a52a487cdbebd Reviewed-on: https://webrtc-review.googlesource.com/43180 Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21809}
2018-01-30 15:33:02 +01:00
"../test:perf_test",
Reland "Switch frame_analyzer to ABSL_FLAG." This is a reland of 3668e9b56198eb8112562d017c47a52050be3561 Original change's description: > Switch frame_analyzer to ABSL_FLAG. > > Bug: webrtc:10616 > Change-Id: I34473632431569406b9086f57caac12434862005 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136801 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28358} TBR=kwiberg@webrtc.org Bug: webrtc:10616 Change-Id: I661ec82938c80964382fcd0692cf9d5cdb132977 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143681 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28361}
2019-06-18 21:09:24 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Replace starts_with and ends_with with Abseil Bug: None Change-Id: I7eae3db1aeb81f0f1d37ff50d5c85c16ecb1f366 Reviewed-on: https://webrtc-review.googlesource.com/c/114221 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26032}
2018-12-13 17:41:25 -08:00
"//third_party/abseil-cpp/absl/strings",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
}
Exclude frame_analyzer_host build on win. No-Try: True Bug: webrtc:11474 Change-Id: If8393410ff0d781c3aa4d5fceebdcc399f77f9e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172585 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30962}
2020-04-01 13:26:24 +02:00
# TODO(bugs.webrtc.org/11474): Enable this on win if needed. For now it
# is only required for Linux and Android.
Exclude frame_analyzer_host build on iOS. Bug: webrtc:11474 Change-Id: I6d5bc2e079330e371e8199d5602b62a0e36fe75a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173721 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31112}
2020-04-20 17:48:26 +02:00
if (!build_with_chromium && !build_with_mozilla && !is_win && !is_ios) {
Do not compile frame_analyzer_host during Chromium builds. Bug: webrtc:9665 Change-Id: I42ff7a02664c3552ea31972a84f1d7d18cab13ac No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100805 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24765}
2018-09-18 13:39:37 +02:00
action("frame_analyzer_host") {
script = "//tools_webrtc/executable_host_build.py"
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
outputs = [ "${root_out_dir}/frame_analyzer_host" ]
Do not compile frame_analyzer_host during Chromium builds. Bug: webrtc:9665 Change-Id: I42ff7a02664c3552ea31972a84f1d7d18cab13ac No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100805 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24765}
2018-09-18 13:39:37 +02:00
args = [
"--executable_name",
"frame_analyzer",
]
}
Reland "Compile frame analyzer for the host machine on perf tests." This is a reland of d8ff3f29ce92e27529e100ecf71afbae6334419f. See https://webrtc-review.googlesource.com/c/src/+/100681/1..4 for the fix. Error "Failed to open video file for emulated camera" should be addressed by that change. Original change's description: > Compile frame analyzer for the host machine on perf tests. > > Bug: webrtc:9665 > Change-Id: I05c01ee4bef0995556b1a679498b3d9132de7c26 > Reviewed-on: https://webrtc-review.googlesource.com/100360 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24756} TBR=phoglund@webrtc.org, oprypin@webrtc.org Bug: webrtc:9665 Change-Id: If6a4f2259dabf50718abf47c9cf303d143a1895a Reviewed-on: https://webrtc-review.googlesource.com/100681 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24762}
2018-09-18 11:03:39 +02:00
}
Make rtp_generator buildable from Chromium. Bug: chromium:942546 Change-Id: I90d077eca55f6cbae119c576d1ba1ec456858377 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168245 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30470}
2020-02-06 10:16:20 +01:00
if (!is_component_build) {
# This target can be built from Chromium but it doesn't support
# is_component_build=true because it depends on WebRTC testonly code
# which is not part of //third_party/webrtc_overrides:webrtc_component.
Add absl_deps field for rtc_test and rtc_executable To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl". Bug: webrtc:12404 Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 10:50:19 +00:00
# Abseil dependencies are not moved to the absl_deps field deliberately.
# If build_with_chromium is true, the absl_deps replaces the dependencies with
# the "//third_party/abseil-cpp:absl" target. Which doesn't include absl/flags
# (and some others) because they cannot be used in Chromiums. Special exception
# for the "frame_analyzer" target in "third_party/abseil-cpp/absl.gni" allows
# it to be build in chromium.
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
rtc_executable("rtp_generator") {
visibility = [ "*" ]
testonly = true
sources = [
"rtp_generator/main.cc",
"rtp_generator/rtp_generator.cc",
"rtp_generator/rtp_generator.h",
]
deps = [
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class Bug: webrtc:10138 Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327 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@{#30033}
2019-12-05 15:59:00 +01:00
"../api:create_frame_generator",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
"../api:transport_api",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Update rtc_tools/rtp_generator to compile This is a trivial CL, updating rtp_generator.cc according to changes in APIs in other places. Bug: webrtc:10807 Change-Id: Ie85c6283f2d78dcf742979378db0b4fb0914c96c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145209 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28526}
2019-07-10 15:34:45 +02:00
"../api/task_queue:default_task_queue_factory",
"../api/task_queue:task_queue",
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../call:fake_network",
"../call:rtp_interfaces",
"../call:rtp_sender",
"../call:simulated_network",
"../call:simulated_packet_receiver",
"../call:video_stream_api",
"../media:rtc_audio_video",
"../media:rtc_media_base",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base",
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_json",
Reland "Refactor rtc_base build targets." 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}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
"../rtc_base/system:file_wrapper",
"../test:fileutils",
"../test:rtp_test_utils",
"../test:video_test_common",
Switch rtp_generator to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I1913292f29191636019d734abf4097d96f372c74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136811 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28366}
2019-06-25 08:46:18 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Add usage message to rtp_generator. Bug: webrtc:10616 Change-Id: Ia13b1e829063986be6b3b4a4b0c194dd4fea3511 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146202 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28609}
2019-07-19 10:02:46 +02:00
"//third_party/abseil-cpp/absl/flags:usage",
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
"//third_party/abseil-cpp/absl/strings",
]
Make rtp_generator buildable from Chromium. Bug: chromium:942546 Change-Id: I90d077eca55f6cbae119c576d1ba1ec456858377 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168245 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30470}
2020-02-06 10:16:20 +01:00
if (build_with_chromium) {
# When building from Chromium, WebRTC's metrics and field trial
# implementations need to be replaced by the Chromium ones.
deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
}
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
}
Move video_replay under rtc_tools/. As pointed out in [1], RTC public tools should live in rtc_tools. [1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50 Bug: chromium:942546 Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 10:02:31 +01:00
# This target can be built from Chromium but it doesn't support
# is_component_build=true because it depends on WebRTC testonly code
# which is not part of //third_party/webrtc_overrides:webrtc_component.
Add absl_deps field for rtc_test and rtc_executable To be able to build these targets in chromium we need to replace all abseil dependencies with "//third_party/abseil-cpp:absl". Bug: webrtc:12404 Change-Id: Ie0f6af73f2abc73e5744520cfd9a6414e2f948e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202762 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33108}
2021-01-29 10:50:19 +00:00
# Abseil dependencies are not moved to the absl_deps field deliberately.
# If build_with_chromium is true, the absl_deps replaces the dependencies with
# the "//third_party/abseil-cpp:absl" target. Which doesn't include absl/flags
# (and some others) because they cannot be used in Chromiums. Special exception
# for the "frame_analyzer" target in "third_party/abseil-cpp/absl.gni" allows
# it to be build in chromium.
Move video_replay under rtc_tools/. As pointed out in [1], RTC public tools should live in rtc_tools. [1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50 Bug: chromium:942546 Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 10:02:31 +01:00
rtc_executable("video_replay") {
visibility = [ "*" ]
testonly = true
sources = [ "video_replay.cc" ]
deps = [
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/transport:field_trial_based_config",
video_replay: add support for IVF file output adding a -decoder_ivf_filename command line flag. BUG=none Change-Id: I895a6bf8093d5f36c17462d97240b17ada4dc9f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201560 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33077}
2021-01-26 18:05:56 +01:00
"../api/video:video_frame",
Move video_replay under rtc_tools/. As pointed out in [1], RTC public tools should live in rtc_tools. [1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50 Bug: chromium:942546 Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 10:02:31 +01:00
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../common_video",
"../media:rtc_internal_video_codecs",
video_replay: add support for IVF file output adding a -decoder_ivf_filename command line flag. BUG=none Change-Id: I895a6bf8093d5f36c17462d97240b17ada4dc9f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201560 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33077}
2021-01-26 18:05:56 +01:00
"../modules/video_coding:video_coding_utility",
Move video_replay under rtc_tools/. As pointed out in [1], RTC public tools should live in rtc_tools. [1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50 Bug: chromium:942546 Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 10:02:31 +01:00
"../rtc_base:checks",
"../rtc_base:rtc_json",
"../rtc_base:stringutils",
"../rtc_base:timeutils",
"../system_wrappers",
"../test:call_config_utils",
"../test:encoder_settings",
"../test:fake_video_codecs",
"../test:null_transport",
"../test:rtp_test_utils",
"../test:run_test",
"../test:run_test_interface",
"../test:test_common",
"../test:test_renderer",
"../test:test_support",
"../test:video_test_common",
"../test:video_test_support",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
]
if (build_with_chromium) {
# When building from Chromium, WebRTC's metrics and field trial
# implementations need to be replaced by the Chromium ones.
deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
}
}
Make rtp_generator buildable from Chromium. Bug: chromium:942546 Change-Id: I90d077eca55f6cbae119c576d1ba1ec456858377 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168245 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30470}
2020-02-06 10:16:20 +01:00
}
Do not build rtp_generator in Chromium builds. This tool is only needed for WebRTC standalone so there is no need to build it on Chromium trybots (if we want to do that, we need to explicitly link against the Chromium's TQ implementation). [1] - https://cs.chromium.org/chromium/src/third_party/webrtc_overrides/BUILD.gn?l=114-124&rcl=3514203635d4f5c2d660784dd3007f1018c9af88 Bug: None Change-Id: Ib75204205717637e6b9b4320deaad5221ce35692 Reviewed-on: https://webrtc-review.googlesource.com/c/121405 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26533}
2019-02-04 15:21:46 +01:00
Make rtp_generator buildable from Chromium. Bug: chromium:942546 Change-Id: I90d077eca55f6cbae119c576d1ba1ec456858377 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168245 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30470}
2020-02-06 10:16:20 +01:00
# Only expose the targets needed by Chromium (e.g. frame_analyzer) to avoid
# building a lot of redundant code as part of Chromium builds.
if (!build_with_chromium) {
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
rtc_executable("psnr_ssim_analyzer") {
Switch psnr_ssim_analyzer to ABSL_FLAG. Bug: webrtc:10616 Change-Id: Ic1a60143c799fc994d8b2411e32ec9435f893906 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143480 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28363}
2019-06-25 08:51:01 +02:00
testonly = true
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
sources = [ "psnr_ssim_analyzer/psnr_ssim_analyzer.cc" ]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
deps = [
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
":video_file_reader",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
":video_quality_analysis",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. 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}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Switch psnr_ssim_analyzer to ABSL_FLAG. Bug: webrtc:10616 Change-Id: Ic1a60143c799fc994d8b2411e32ec9435f893906 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143480 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28363}
2019-06-25 08:51:01 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Add usage message to psnr_ssim_analyzer. Bug: webrtc:10616 Change-Id: I157fbf63b3794a1f655901ace7355edb1240cc72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146203 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28610}
2019-07-19 10:08:41 +02:00
"//third_party/abseil-cpp/absl/flags:usage",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
}
Use source_sets in component builds and static_library in release builds. 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}
2019-10-17 21:32:04 +02:00
rtc_library("reference_less_video_analysis_lib") {
Extract an interface from the perf results logger. 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}
2020-01-13 15:10:40 +01:00
testonly = true
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
sources = [
"frame_analyzer/reference_less_video_analysis_lib.cc",
"frame_analyzer/reference_less_video_analysis_lib.h",
]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
deps = [
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
":video_file_reader",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
":video_quality_analysis",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. 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}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
]
}
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
rtc_executable("reference_less_video_analysis") {
Switch reference_less_video_analysis to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I7b8b1cdae64d8d704056c19806ff8d4708810210 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28364}
2019-06-25 08:49:03 +02:00
testonly = true
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
sources = [ "frame_analyzer/reference_less_video_analysis.cc" ]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
deps = [
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
":reference_less_video_analysis_lib",
Switch reference_less_video_analysis to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I7b8b1cdae64d8d704056c19806ff8d4708810210 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28364}
2019-06-25 08:49:03 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Add usage message to reference_less_video_analysis. Bug: webrtc:10616 Change-Id: Ibbaba9ec6a74466021e70d01f3a8ff4c3fd212b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146205 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28611}
2019-07-19 10:32:16 +02:00
"//third_party/abseil-cpp/absl/flags:usage",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
}
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
rtc_executable("rgba_to_i420_converter") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Switch rgba_to_i420_converter to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I45f94e0f43e84b041c72b1bdc5b35ee0e970397e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143179 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28362}
2019-06-25 08:52:47 +02:00
testonly = true
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
sources = [
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
"converter/converter.cc",
"converter/converter.h",
"converter/rgba_to_i420_converter.cc",
]
deps = [
"../common_video",
Switch rgba_to_i420_converter to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I45f94e0f43e84b041c72b1bdc5b35ee0e970397e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143179 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28362}
2019-06-25 08:52:47 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Add usage message to rgba_to_i420_converter. Bug: webrtc:10616 Change-Id: Id8268bada4ec42130d9b7b105da73218dfa43519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146209 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28614}
2019-07-19 11:58:07 +02:00
"//third_party/abseil-cpp/absl/flags:usage",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
}
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
if (rtc_enable_protobuf) {
proto_library("chart_proto") {
Make the event log visualizer chart proto publicly visible. Bug: None Change-Id: I9b23216e8f81e7cf7ca422d2b5c573e450f4df2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134108 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27743}
2019-04-24 17:12:45 +02:00
visibility = [ "*" ]
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
sources = [
Add rtc_ prefix to the event_log_visualizer directory. No-Try: True Bug: None Change-Id: Iaa2b273ddab6567321f11bf74a91751cbdf957a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146710 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28681}
2019-07-25 13:57:41 +02:00
"rtc_event_log_visualizer/proto/chart.proto",
"rtc_event_log_visualizer/proto/chart_enums.proto",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
]
Add rtc_ prefix to the event_log_visualizer directory. No-Try: True Bug: None Change-Id: Iaa2b273ddab6567321f11bf74a91751cbdf957a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146710 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28681}
2019-07-25 13:57:41 +02:00
proto_out_dir = "rtc_tools/rtc_event_log_visualizer/proto"
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
}
Use source_sets in component builds and static_library in release builds. 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}
2019-10-17 21:32:04 +02:00
rtc_library("event_log_visualizer_utils") {
Revert "Revert "Make some more targets publicly visible"" This reverts commit 55d1809d0d73592a1ddf4f0fb02ce7444fa066aa. Reason for revert: This cl was not the culprit for breaking chrome content/renderer deps. Original change's description: > Revert "Make some more targets publicly visible" > > This reverts commit 60d179256213c7516808aff827637cab8a47de89. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > Make some more targets publicly visible > > > > To fix build errors introduced by > > https://webrtc-review.googlesource.com/c/src/+/24140 > > > > BUG=webrtc:8254 > > NOTRY=true > > > > Change-Id: I9cdf9cee39735368af78291134dbad70aebb7195 > > Reviewed-on: https://webrtc-review.googlesource.com/38660 > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21552} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I475ac382218fa77d33abc595f0773275d715a28e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38740 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21554} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: If8e9b7b1c934ec4b5ed61941c845e62e43bef97e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38841 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21559}
2018-01-10 15:57:32 +00:00
visibility = [ "*" ]
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
sources = [
Refactor/reimplement RTC event log triage alerts. - Moves AnalyzerConfig and helper functions IsAudioSsrc, IsVideoSsrc, IsRtxSsrc, GetStreamNam and GetLayerName to analyzer_common.h - Moves log_segments() code to rtc_event_log_parser.h - Moves TriageAlert/Notification code to a new file with a couple of minor fixes to make it less spammy. Bug: webrtc:11566 Change-Id: Ib33941d8185f7382fc72ed65768e46015e0320de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174824 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31318}
2020-05-19 10:57:24 +02:00
"rtc_event_log_visualizer/alerts.cc",
"rtc_event_log_visualizer/alerts.h",
Move NetEq and ANA plotting to a separate file. Bug: webrtc:11566 Change-Id: I6d6176ff72a158a1629e14b539de2e928e7d02a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176510 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@google.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31472}
2020-06-05 10:47:19 +02:00
"rtc_event_log_visualizer/analyze_audio.cc",
"rtc_event_log_visualizer/analyze_audio.h",
Add rtc_ prefix to the event_log_visualizer directory. No-Try: True Bug: None Change-Id: Iaa2b273ddab6567321f11bf74a91751cbdf957a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146710 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28681}
2019-07-25 13:57:41 +02:00
"rtc_event_log_visualizer/analyzer.cc",
"rtc_event_log_visualizer/analyzer.h",
Refactor/reimplement RTC event log triage alerts. - Moves AnalyzerConfig and helper functions IsAudioSsrc, IsVideoSsrc, IsRtxSsrc, GetStreamNam and GetLayerName to analyzer_common.h - Moves log_segments() code to rtc_event_log_parser.h - Moves TriageAlert/Notification code to a new file with a couple of minor fixes to make it less spammy. Bug: webrtc:11566 Change-Id: Ib33941d8185f7382fc72ed65768e46015e0320de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174824 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31318}
2020-05-19 10:57:24 +02:00
"rtc_event_log_visualizer/analyzer_common.cc",
"rtc_event_log_visualizer/analyzer_common.h",
Add rtc_ prefix to the event_log_visualizer directory. No-Try: True Bug: None Change-Id: Iaa2b273ddab6567321f11bf74a91751cbdf957a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146710 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28681}
2019-07-25 13:57:41 +02:00
"rtc_event_log_visualizer/log_simulation.cc",
"rtc_event_log_visualizer/log_simulation.h",
"rtc_event_log_visualizer/plot_base.cc",
"rtc_event_log_visualizer/plot_base.h",
"rtc_event_log_visualizer/plot_protobuf.cc",
"rtc_event_log_visualizer/plot_protobuf.h",
"rtc_event_log_visualizer/plot_python.cc",
"rtc_event_log_visualizer/plot_python.h",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
]
deps = [
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
":chart_proto",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../api:function_view",
Cleanup of bwe_defines.h Delete unused macros BWE_MIN and BWE_MAX. Move enum RateControlState: Make it a private enum class in AimdRateControl, the only user. Change users of the header file that only need BandwidthUsage, to instead include api/network_state_predictor.h, the file defining this class. As a result, fewer dependencies on modules/remote_bitrate_estimator. Bug: None Change-Id: I4450c79dd58e1875d64dd74d1ae2cb7d911a14b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195222 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32704}
2020-11-24 17:50:31 +01:00
"../api:network_state_predictor_api",
Deprecate inheritance hierachy of plot formats in event_log_visualizer. Instead add separate printing functions for each plot format in the base class. Bug: webrtc:11566 Change-Id: I8adfc983b4e8a66c477de4022c2d97b6975d7e5c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176563 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31496}
2020-06-09 17:17:21 +02:00
"../rtc_base:deprecation",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../rtc_base:ignore_wundef",
Reland "Add AudioDecoderFactory to NetEqTest constructor." This is a reland of daa970f33e1923c5651a4a63c18e3d5361d0a795 Original change's description: > Add AudioDecoderFactory to NetEqTest constructor. > > Update EventLogAnalyzer to not depend on builtin audio decoders. > > Bug: webrtc:8396, webrtc:10080 > Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738 > Reviewed-on: https://webrtc-review.googlesource.com/c/114301 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26026} Tbr: kwiberg@webrtc.org Bug: webrtc:8396, webrtc:10080 Change-Id: I598ce1cd41676b1992b0973b09476eeeb0e602d2 Reviewed-on: https://webrtc-review.googlesource.com/c/114940 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26058}
2018-12-19 15:06:17 +01:00
# TODO(kwiberg): Remove this dependency.
"../api/audio_codecs:audio_codecs_api",
Adds WebRtcKeyValueConfig interface 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}
2019-01-17 16:24:12 +01:00
"../api/transport:field_trial_based_config",
Reland "Add AudioDecoderFactory to NetEqTest constructor." This is a reland of daa970f33e1923c5651a4a63c18e3d5361d0a795 Original change's description: > Add AudioDecoderFactory to NetEqTest constructor. > > Update EventLogAnalyzer to not depend on builtin audio decoders. > > Bug: webrtc:8396, webrtc:10080 > Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738 > Reviewed-on: https://webrtc-review.googlesource.com/c/114301 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26026} Tbr: kwiberg@webrtc.org Bug: webrtc:8396, webrtc:10080 Change-Id: I598ce1cd41676b1992b0973b09476eeeb0e602d2 Reviewed-on: https://webrtc-review.googlesource.com/c/114940 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26058}
2018-12-19 15:06:17 +01:00
"../api/transport:goog_cc",
Add log based GoogCC simulation to visualizer. 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}
2019-03-13 08:56:58 +01:00
"../api/transport:network_control",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
"../call:call_interfaces",
"../call:video_stream_api",
"../logging:rtc_event_log_parser",
Reland "Break up rtc_event_log_api to solve circular dependencies." This is a reland of 001546da953275c7a39eb220592b440c9b47d756 Original change's description: > Break up rtc_event_log_api to solve circular dependencies. > > The original rtc_event_log_api is refactored to a pure API target plus > multiple targets coupled with WebRTC implementations. > > Bug: None > Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f > Reviewed-on: https://webrtc-review.googlesource.com/43247 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Qingsi Wang <qingsi@google.com> > Cr-Commit-Position: refs/heads/master@{#21811} TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21862}
2018-02-01 11:04:46 -08:00
"../logging:rtc_stream_config",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
"../modules/audio_coding:ana_debug_dump_proto",
Stop using public_deps in logging/. Bug: webrtc:8603 Change-Id: Id0df997620a27e47067e4b21e4e8db16aec90640 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30940 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21228}
2017-12-12 12:46:13 +01:00
"../modules/audio_coding:audio_network_adaptor",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
"../modules/audio_coding:neteq_tools",
"../modules/congestion_controller",
Removes redundant delay based bwe. This removes the legacy DelayBasedBwe to reduce code redundancy and avoid the risk of applying changes on only one version. Bug: webrtc:8415 Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a Reviewed-on: https://webrtc-review.googlesource.com/85364 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23798}
2018-07-02 09:25:25 +02:00
"../modules/congestion_controller/goog_cc:delay_based_bwe",
Replaces redundant congestion controller components This CL replaces components in the congestion controller module that are identical to equivalent components in the rtp and goog_cc subfolder. Some redundant components are left as they were not trivial to replace. Bug: webrtc:8415 Change-Id: I86a1f164d7b100b8ec8ba7dbc1c9bda2128a4f37 Reviewed-on: https://webrtc-review.googlesource.com/78521 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23384}
2018-05-24 14:17:06 +02:00
"../modules/congestion_controller/goog_cc:estimators",
Removes analyzer dependency on legacy congestion controller. Bug: webrtc:9586 Change-Id: Ic1f2445d6432202aeba9164acd49b75261e91aa0 Reviewed-on: https://webrtc-review.googlesource.com/c/105107 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25183}
2018-10-15 18:46:51 +02:00
"../modules/congestion_controller/rtp:transport_feedback",
Delete deprecated constructor of SendSideCongestionController. Move packet_router #include to where it's needed, and delete unused MockPacketRouter. Bug: webrtc:6847 Change-Id: I03c86c6fb8b413f5a535a237fa1724cc10960ffa Reviewed-on: https://webrtc-review.googlesource.com/17320 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20564}
2017-10-31 10:19:10 +01:00
"../modules/pacing",
Removes analyzer dependency on legacy congestion controller. Bug: webrtc:9586 Change-Id: Ic1f2445d6432202aeba9164acd49b75261e91aa0 Reviewed-on: https://webrtc-review.googlesource.com/c/105107 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25183}
2018-10-15 18:46:51 +02:00
"../modules/remote_bitrate_estimator",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
"../modules/rtp_rtcp",
Reland "Add AudioDecoderFactory to NetEqTest constructor." This is a reland of daa970f33e1923c5651a4a63c18e3d5361d0a795 Original change's description: > Add AudioDecoderFactory to NetEqTest constructor. > > Update EventLogAnalyzer to not depend on builtin audio decoders. > > Bug: webrtc:8396, webrtc:10080 > Change-Id: Ie02ed9cda6d4f11bfdf2e65eb6482283b7520738 > Reviewed-on: https://webrtc-review.googlesource.com/c/114301 > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26026} Tbr: kwiberg@webrtc.org Bug: webrtc:8396, webrtc:10080 Change-Id: I598ce1cd41676b1992b0973b09476eeeb0e602d2 Reviewed-on: https://webrtc-review.googlesource.com/c/114940 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26058}
2018-12-19 15:06:17 +01:00
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_numerics",
"../rtc_base:stringutils",
Add simulation of robust throughput estimator to the event log analyzer Bug: webrtc:11566 Change-Id: I873d1c1bd6682a973b3a130289390e09ef47cc37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177017 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31538}
2020-06-17 11:55:55 +02:00
"../test:explicit_key_value_config",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Plot delay feedback in RTCP arrival order. This fixes a minor bug in the event_log_visualizer where packets are processed in RTP send time order rather than RTCP arrival time order. The bug makes time appear to move backwards if RTCP feedback for a later RTP packet arrives before the feedback of an earlier RTP packet. Bug: None Change-Id: I06e8a25d5c65602bedcfd9e4ea1d23874bee9318 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156169 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29448}
2019-10-11 15:06:58 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
Avoid depending on testonly target in event_log_visualizer_utils. This is done by creating a custom ReplacementAudioDecoderFactory. Bug: webrtc:8396, webrtc:10080 Change-Id: Ie1cb614fec855b82d65c6ef86c3593f547254559 Reviewed-on: https://webrtc-review.googlesource.com/c/116795 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26217}
2019-01-11 13:01:29 +01:00
"//third_party/abseil-cpp/absl/strings",
Move NetEq and ANA plotting to a separate file. Bug: webrtc:11566 Change-Id: I6d6176ff72a158a1629e14b539de2e928e7d02a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176510 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@google.com> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31472}
2020-06-05 10:47:19 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Expose less targets in webrtc/rtc_tools in Chromium builds. This should solve recent linking problems with frame_editor and also speed up the Chromium build a little. BUG=None NOTRY=True Review-Url: https://codereview.webrtc.org/3006093002 Cr-Commit-Position: refs/heads/master@{#19677}
2017-09-04 23:42:45 -07:00
]
}
}
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
}
if (rtc_include_tests) {
Add YUV to IVF video converter util Bug: webrtc:10138 Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 15:03:45 +01:00
rtc_executable("yuv_to_ivf_converter") {
visibility = [ "*" ]
testonly = true
Exclude frame_analyzer_host build on win. No-Try: True Bug: webrtc:11474 Change-Id: If8393410ff0d781c3aa4d5fceebdcc399f77f9e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172585 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30962}
2020-04-01 13:26:24 +02:00
sources = [ "converter/yuv_to_ivf_converter.cc" ]
Add YUV to IVF video converter util Bug: webrtc:10138 Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 15:03:45 +01:00
deps = [
"../api:create_frame_generator",
"../api:frame_generator_api",
"../api/task_queue:default_task_queue_factory",
"../api/video:encoded_image",
"../api/video:video_frame",
"../api/video_codecs:video_codecs_api",
"../media:rtc_media_base",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
"../rtc_base:checks",
"../rtc_base:criticalsection",
"../rtc_base:logging",
"../rtc_base:rtc_event",
"../rtc_base:rtc_task_queue",
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: Ia85cf457ac16ebec0a70d319506415e4dec46f30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178903 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31693}
2020-07-08 23:04:37 +02:00
"../rtc_base/synchronization:mutex",
Add YUV to IVF video converter util Bug: webrtc:10138 Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 15:03:45 +01:00
"../rtc_base/system:file_wrapper",
"../test:video_test_common",
"../test:video_test_support",
"//third_party/abseil-cpp/absl/debugging:failure_signal_handler",
"//third_party/abseil-cpp/absl/debugging:symbolize",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/strings",
]
}
Harmonize paths to dependent targets. 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}
2018-10-30 21:12:42 +01:00
if (rtc_enable_protobuf && !build_with_chromium) {
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
rtc_executable("event_log_visualizer") {
testonly = true
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
sources = [ "rtc_event_log_visualizer/main.cc" ]
Add data dependency to event_log_visualizer. This .wav file is an implicit data dependency, this CL adds this information to the build system. Bug: None Change-Id: Ia953e63d4658debce3cecb93bb1f3e749fe52f54 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166044 Reviewed-by: Ali Tofigh <alito@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30257}
2020-01-14 19:20:50 +01:00
data = [
# If --wav_filename is not provided, event_log_visualizer uses
# EN_script2_F_sp2_B1.wav by default. This is a good default to use
# for example with flags --plot=all when there is no need to use a
# specific .wav file.
"../resources/audio_processing/conversational_speech/EN_script2_F_sp2_B1.wav",
]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
deps = [
":event_log_visualizer_utils",
Move NetEq headers to api/ This CL also introduces NetEqFactory and NetEqControllerFactory interfaces, as well as several convenience classes for working with them: DefaultNetEqFactory, DefaultNetEqControllerFactory and CustomNetEqFactory. Bug: webrtc:11005 Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29671}
2019-10-31 14:38:11 +01:00
"../api/neteq:neteq_api",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Stop using public_deps in rtc_tools. Bug: webrtc:8603 Change-Id: I8126971f80310ad077a1e536c022a296612f573a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30300 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21162}
2017-12-06 16:31:18 +01:00
"../logging:rtc_event_log_parser",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../modules/audio_coding:neteq",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
Stop using public_deps in logging/. Bug: webrtc:8603 Change-Id: Id0df997620a27e47067e4b21e4e8db16aec90640 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/30940 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21228}
2017-12-12 12:46:13 +01:00
"../rtc_base:protobuf_utils",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
Remove deprecated field_trial_default and metrics_default. 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}
2018-09-28 08:51:10 +02:00
"../system_wrappers:field_trial",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../test:fileutils",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"../test:test_support",
Add string-based IDs for event log visualizer graphs and update command line flags. 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}
2019-06-11 17:39:38 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
Add usage message to event_log_visualizer. Bug: webrtc:10616 Change-Id: I1623175abc7e60c1a3cf29e3d99fdde0e4140b1f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146208 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28663}
2019-07-24 15:29:47 +02:00
"//third_party/abseil-cpp/absl/flags:config",
Switch event_log_visualizer to ABSL_FLAG. Bug: webrtc:10616 Change-Id: I915ce581b082e081501f4e8545e939a81baff366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143968 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28575}
2019-07-16 11:55:08 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Add usage message to event_log_visualizer. Bug: webrtc:10616 Change-Id: I1623175abc7e60c1a3cf29e3d99fdde0e4140b1f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146208 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28663}
2019-07-24 15:29:47 +02:00
"//third_party/abseil-cpp/absl/flags:usage",
"//third_party/abseil-cpp/absl/strings",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
}
}
tools_unittests_resources = [
Reland "Add tool for aliging video files" This is a reland of b2c0e8f60fad10e2786e5e131136a0da1299d883 Original change's description: > Add tool for aliging video files > > This class adds logic for aligning a test video to a reference video > by an algorithm that maximizes SSIM between them. Aligned videos will be > easier to run video quality metrics on. This is a generic way of > aligning videos and can be replace the intrusive barcode stamping that > we currently use. This will be done in a follow-up CL. > > Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94773 > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24407} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: I35d6b0e598335b8d80fbfa37ba06d5c651bda4f6 Reviewed-on: https://webrtc-review.googlesource.com/98040 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24580}
2018-09-05 15:04:08 +02:00
"../resources/foreman_128x96.yuv",
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"../resources/foreman_cif.yuv",
"../resources/reference_less_video_test_file.y4m",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
if (is_ios) {
bundle_data("tools_unittests_bundle_data") {
testonly = true
sources = tools_unittests_resources
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
}
}
rtc_test("tools_unittests") {
testonly = true
sources = [
Add tool for aligning color space of video files 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}
2018-09-11 12:56:06 +02:00
"frame_analyzer/linear_least_squares_unittest.cc",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"frame_analyzer/reference_less_video_analysis_unittest.cc",
Add tool for aligning color space of video files 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}
2018-09-11 12:56:06 +02:00
"frame_analyzer/video_color_aligner_unittest.cc",
Add tool for aligning cropped region of video files 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}
2018-11-22 15:17:30 +01:00
"frame_analyzer/video_geometry_aligner_unittest.cc",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"frame_analyzer/video_quality_analysis_unittest.cc",
Reland "Add tool for aliging video files" This is a reland of b2c0e8f60fad10e2786e5e131136a0da1299d883 Original change's description: > Add tool for aliging video files > > This class adds logic for aligning a test video to a reference video > by an algorithm that maximizes SSIM between them. Aligned videos will be > easier to run video quality metrics on. This is a generic way of > aligning videos and can be replace the intrusive barcode stamping that > we currently use. This will be done in a follow-up CL. > > Change-Id: I71cf1e2179c0f1e03eff9e4d8fc492fd5cfbbb1c > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94773 > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24407} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: I35d6b0e598335b8d80fbfa37ba06d5c651bda4f6 Reviewed-on: https://webrtc-review.googlesource.com/98040 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24580}
2018-09-05 15:04:08 +02:00
"frame_analyzer/video_temporal_aligner_unittest.cc",
Add a check that sanitizers cause a fatal failure Inspired by https://chromium.googlesource.com/chromium/src/+/master/base/tools_sanity_unittest.cc BUG=webrtc:8214 R=kjellander@webrtc.org, kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3005273002 . Cr-Commit-Position: refs/heads/master@{#19830}
2017-09-14 13:05:00 +02:00
"sanitizers_unittest.cc",
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
"video_file_reader_unittest.cc",
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
"video_file_writer_unittest.cc",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
deps = [
Reland "Add Y4mFileReader" This is a reland of 404be7f302358e6be16aadeba8bc8f8aba348c0f It adds support for reading .yuv files as well to not break anything. Original change's description: > Add Y4mFileReader > > Encapsulate logic for reading .y4m video files in a single class. We > currently have spread out logic for opening .y4m files with partial > parsing. This CL consolidates this logic into a single class with a well > defined interface. > > Change-Id: Id61673b3c95a0053b30e95b4cf382e1c6b05fc30 > Bug: webrtc:9642 > Reviewed-on: https://webrtc-review.googlesource.com/94772 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Paulina Hensman <phensman@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#24398} TBR=phensman,phoglund Bug: webrtc:9642 Change-Id: Idecc5ec5da767221a5f5b439989f4fe07e3b3615 Reviewed-on: https://webrtc-review.googlesource.com/97983 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24571}
2018-09-05 10:46:18 +02:00
":video_file_reader",
Add WriteVideoToFile to video_file_reader. 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}
2018-09-14 11:32:00 +02:00
":video_file_writer",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
":video_quality_analysis",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. 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}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../common_video",
Add a check that sanitizers cause a fatal failure Inspired by https://chromium.googlesource.com/chromium/src/+/master/base/tools_sanity_unittest.cc BUG=webrtc:8214 R=kjellander@webrtc.org, kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/3005273002 . Cr-Commit-Position: refs/heads/master@{#19830}
2017-09-14 13:05:00 +02:00
"../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Reland "Refactor rtc_base build targets." 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}
2021-01-15 10:41:01 +01:00
"../rtc_base:null_socket_server",
"../rtc_base:threading",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../test:fileutils",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"../test:test_main",
Harmonize paths to dependent targets. 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}
2018-10-30 21:12:42 +01:00
"../test:test_support",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
"//testing/gtest",
Add tool for aligning color space of video files 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}
2018-09-11 12:56:06 +02:00
"//third_party/libyuv",
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
]
Harmonize paths to dependent targets. 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}
2018-10-30 21:12:42 +01:00
if (!build_with_chromium) {
Remove rtc_tools/frame_editing. This tool is unused, this CL removes it in order to reduce the cost of the maintenance (in the last 2 years only maintenance commits have been landed in this directory). Bug: None Change-Id: Ieec113bc25c480405d32e284a0456572758352e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146204 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28619}
2019-07-19 11:35:45 +02:00
deps += [ ":reference_less_video_analysis_lib" ]
Harmonize paths to dependent targets. 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}
2018-10-30 21:12:42 +01:00
}
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
if (rtc_enable_protobuf) {
deps += [ "network_tester:network_tester_unittests" ]
}
data = tools_unittests_resources
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
if (is_ios) {
deps += [ ":tools_unittests_bundle_data" ]
}
}
if (rtc_enable_protobuf) {
Move audioproc_f to rtc_tools. The motivation in https://webrtc-review.googlesource.com/c/src/+/32340/3 applies here as well. We would like to use this tool downstream. Bug: None Change-Id: Id5b23f792679ab9c07294bfb8e53119c423044b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161681 Commit-Queue: Daniel Johansson <dajo@webrtc.org> Reviewed-by: Daniel Johansson <dajo@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30051}
2019-12-10 11:00:38 +01:00
rtc_executable("audioproc_f") {
testonly = true
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
sources = [ "audioproc_f/audioproc_float_main.cc" ]
Move audioproc_f to rtc_tools. The motivation in https://webrtc-review.googlesource.com/c/src/+/32340/3 applies here as well. We would like to use this tool downstream. Bug: None Change-Id: Id5b23f792679ab9c07294bfb8e53119c423044b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161681 Commit-Queue: Daniel Johansson <dajo@webrtc.org> Reviewed-by: Daniel Johansson <dajo@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30051}
2019-12-10 11:00:38 +01:00
deps = [
"../api:audioproc_f_api",
"../modules/audio_processing",
"../modules/audio_processing:api",
"../rtc_base:rtc_base_approved",
]
}
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
copy("rtp_analyzer") {
sources = [
"py_event_log_analyzer/misc.py",
"py_event_log_analyzer/pb_parse.py",
"py_event_log_analyzer/rtp_analyzer.py",
"py_event_log_analyzer/rtp_analyzer.sh",
]
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
outputs = [ "$root_build_dir/{{source_file_part}}" ]
deps = [ "../logging:rtc_event_log_proto" ]
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
} # rtp_analyzer
rtc_executable("unpack_aecdump") {
Making rtc_tools:unpack_aecdump publicly visible. Bug: webrtc:8254 Change-Id: I0dce33dc45f108e5dcfe18e22480d533f93073c4 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/38960 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21569}
2018-01-10 22:18:27 +01:00
visibility = [ "*" ]
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
testonly = true
Reformat GN files. `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}
2020-01-21 12:10:10 +01:00
sources = [ "unpack_aecdump/unpack.cc" ]
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
deps = [
Make unpack_aecdump unpack RuntimeSettings 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}
2019-05-21 12:23:31 +02:00
"../api:function_view",
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
"../common_audio",
"../modules/audio_processing",
"../modules/audio_processing:audioproc_debug_proto",
"../modules/audio_processing:audioproc_debug_proto",
"../modules/audio_processing:audioproc_protobuf_utils",
"../modules/audio_processing:audioproc_test_utils",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../rtc_base:ignore_wundef",
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
Switch unpack_aecdump to ABSL_FLAG. Bug: webrtc:10616 Change-Id: Ia6d0104d589b4b966ad6705493d88d3cb701c266 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144600 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28486}
2019-07-04 10:19:10 +02:00
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
Move unpack_aecdump to a more public location. This tool is used downstream, so we want to christen rtc_tools as a kind of api dir for tools. Tools in other locations should be considered off limits. I chose rtc_tools because video_quality_toolchain is already there, which is also used downstream. Bug: None Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836 Reviewed-on: https://webrtc-review.googlesource.com/32340 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21236}
2017-12-12 15:53:31 +01:00
]
} # unpack_aecdump
Move webrtc/{tools => rtc_tools} Leaving compatibility script in webrtc/tools/compare_videos.py to avoid breaking our video quality tests in Chromium. Forwarding GN targets are left in webrtc/tools/BUILD.gn. BUG=webrtc:7855 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2965593002 Cr-Commit-Position: refs/heads/master@{#18848}
2017-06-30 03:04:59 -07:00
}
}
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