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diff --git a/third_party/libwebrtc/rtc_tools/BUILD.gn b/third_party/libwebrtc/rtc_tools/BUILD.gn
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+++ b/third_party/libwebrtc/rtc_tools/BUILD.gn
@@ -0,0 +1,600 @@
+# 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")
+if (rtc_enable_protobuf) {
+ import("//third_party/protobuf/proto_library.gni")
+}
+
+group("rtc_tools") {
+ # This target shall build all targets in tools/.
+ testonly = true
+
+ deps = [ ":video_file_reader" ]
+ if (!build_with_chromium) {
+ deps += [
+ ":frame_analyzer",
+ ":psnr_ssim_analyzer",
+ ":video_quality_analysis",
+ ]
+ }
+ if (!build_with_chromium && rtc_enable_protobuf) {
+ deps += [ ":chart_proto" ]
+ }
+ if (!build_with_chromium && rtc_include_tests) {
+ deps += [ ":tools_unittests" ]
+ }
+ if (rtc_include_tests && rtc_enable_protobuf) {
+ deps += [
+ ":rtp_analyzer",
+ "network_tester",
+ ]
+ }
+ if (rtc_include_tests && rtc_enable_protobuf && !build_with_chromium) {
+ deps += [
+ ":audioproc_f",
+ ":event_log_visualizer",
+ ":rtc_event_log_to_text",
+ ":unpack_aecdump",
+ ]
+ }
+ if (!build_with_chromium && rtc_enable_grpc) {
+ deps += [ "data_channel_benchmark" ]
+ }
+}
+
+rtc_library("video_file_reader") {
+ sources = [
+ "video_file_reader.cc",
+ "video_file_reader.h",
+ ]
+ deps = [
+ "../api:make_ref_counted",
+ "../api:scoped_refptr",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "../rtc_base:refcount",
+ "../rtc_base:stringutils",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+}
+
+rtc_library("video_file_writer") {
+ sources = [
+ "video_file_writer.cc",
+ "video_file_writer.h",
+ ]
+ deps = [
+ ":video_file_reader",
+ "../api:scoped_refptr",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ "../rtc_base:logging",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+}
+
+rtc_library("video_quality_analysis") {
+ testonly = true
+ sources = [
+ "frame_analyzer/linear_least_squares.cc",
+ "frame_analyzer/linear_least_squares.h",
+ "frame_analyzer/video_color_aligner.cc",
+ "frame_analyzer/video_color_aligner.h",
+ "frame_analyzer/video_geometry_aligner.cc",
+ "frame_analyzer/video_geometry_aligner.h",
+ "frame_analyzer/video_quality_analysis.cc",
+ "frame_analyzer/video_quality_analysis.h",
+ "frame_analyzer/video_temporal_aligner.cc",
+ "frame_analyzer/video_temporal_aligner.h",
+ ]
+ deps = [
+ ":video_file_reader",
+ "../api:array_view",
+ "../api:make_ref_counted",
+ "../api:scoped_refptr",
+ "../api/numerics",
+ "../api/test/metrics:metric",
+ "../api/test/metrics:metrics_logger",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ "../common_video",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "//third_party/libyuv",
+ ]
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+# TODO(bugs.webrtc.org/11474): Enable this on win if needed. For now it
+# is only required for Linux and Android.
+if (!build_with_chromium && !build_with_mozilla && !is_win && !is_ios) {
+ action("frame_analyzer_host") {
+ script = "//tools_webrtc/executable_host_build.py"
+ outputs = [ "${root_out_dir}/frame_analyzer_host" ]
+ args = [
+ "--executable_name",
+ "frame_analyzer",
+ ]
+ }
+}
+
+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.
+
+ # 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.
+ rtc_executable("frame_analyzer") {
+ visibility = [ "*" ]
+ testonly = true
+ sources = [ "frame_analyzer/frame_analyzer.cc" ]
+
+ deps = [
+ ":video_file_reader",
+ ":video_file_writer",
+ ":video_quality_analysis",
+ "../api:make_ref_counted",
+ "../api:scoped_refptr",
+ "../api/test/metrics:chrome_perf_dashboard_metrics_exporter",
+ "../api/test/metrics:global_metrics_logger_and_exporter",
+ "../api/test/metrics:metrics_exporter",
+ "../api/test/metrics:stdout_metrics_exporter",
+ "../rtc_base:stringutils",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ "//third_party/abseil-cpp/absl/strings",
+ ]
+
+ 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" ]
+ }
+ }
+
+ # 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.
+
+ # 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 "rtp_generator" target in "third_party/abseil-cpp/absl.gni" allows
+ # it to be build in chromium.
+ rtc_executable("rtp_generator") {
+ visibility = [ "*" ]
+ testonly = true
+ sources = [
+ "rtp_generator/main.cc",
+ "rtp_generator/rtp_generator.cc",
+ "rtp_generator/rtp_generator.h",
+ ]
+
+ deps = [
+ "../api:create_frame_generator",
+ "../api:rtp_parameters",
+ "../api:transport_api",
+ "../api/rtc_event_log",
+ "../api/task_queue:default_task_queue_factory",
+ "../api/task_queue:task_queue",
+ "../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:media_constants",
+ "../media:rtc_audio_video",
+ "../media:rtc_media_base",
+ "../rtc_base:rtc_json",
+ "../rtc_base:threading",
+ "../rtc_base/system:file_wrapper",
+ "../test:fileutils",
+ "../test:rtp_test_utils",
+ "../test:video_test_common",
+ "../video/config:encoder_config",
+ "../video/config:streams_config",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ "//third_party/abseil-cpp/absl/flags:usage",
+ "//third_party/abseil-cpp/absl/strings",
+ ]
+ 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" ]
+ }
+ }
+
+ # 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.
+
+ # 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 "video_replay" target in "third_party/abseil-cpp/absl.gni" allows
+ # it to be build in chromium.
+ rtc_executable("video_replay") {
+ visibility = [ "*" ]
+ testonly = true
+ sources = [ "video_replay.cc" ]
+ deps = [
+ "../api:field_trials",
+ "../api:rtp_parameters",
+ "../api/rtc_event_log",
+ "../api/task_queue:default_task_queue_factory",
+ "../api/test/video:function_video_factory",
+ "../api/transport:field_trial_based_config",
+ "../api/units:timestamp",
+ "../api/video:video_frame",
+ "../api/video_codecs:video_codecs_api",
+ "../call",
+ "../call:call_interfaces",
+ "../common_video",
+ "../media:rtc_internal_video_codecs",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../modules/video_coding:video_coding_utility",
+ "../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_loop",
+ "../test:run_test",
+ "../test:run_test_interface",
+ "../test:test_common",
+ "../test:test_renderer",
+ "../test:test_support",
+ "../test:video_test_common",
+ "../test:video_test_support",
+ "../test/time_controller:time_controller",
+ "//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" ]
+ }
+ }
+}
+
+# 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) {
+ rtc_executable("psnr_ssim_analyzer") {
+ testonly = true
+ sources = [ "psnr_ssim_analyzer/psnr_ssim_analyzer.cc" ]
+
+ deps = [
+ ":video_file_reader",
+ ":video_quality_analysis",
+ "../api:scoped_refptr",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ "//third_party/abseil-cpp/absl/flags:usage",
+ ]
+ }
+
+ rtc_library("reference_less_video_analysis_lib") {
+ testonly = true
+ sources = [
+ "frame_analyzer/reference_less_video_analysis_lib.cc",
+ "frame_analyzer/reference_less_video_analysis_lib.h",
+ ]
+
+ deps = [
+ ":video_file_reader",
+ ":video_quality_analysis",
+ "../api:scoped_refptr",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ ]
+ }
+
+ rtc_executable("reference_less_video_analysis") {
+ testonly = true
+ sources = [ "frame_analyzer/reference_less_video_analysis.cc" ]
+
+ deps = [
+ ":reference_less_video_analysis_lib",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ "//third_party/abseil-cpp/absl/flags:usage",
+ ]
+ }
+
+ if (rtc_enable_protobuf) {
+ proto_library("chart_proto") {
+ visibility = [ "*" ]
+ sources = [
+ "rtc_event_log_visualizer/proto/chart.proto",
+ "rtc_event_log_visualizer/proto/chart_enums.proto",
+ ]
+ proto_out_dir = "rtc_tools/rtc_event_log_visualizer/proto"
+ }
+
+ rtc_library("event_log_visualizer_utils") {
+ visibility = [ "*" ]
+ sources = [
+ "rtc_event_log_visualizer/alerts.cc",
+ "rtc_event_log_visualizer/alerts.h",
+ "rtc_event_log_visualizer/analyze_audio.cc",
+ "rtc_event_log_visualizer/analyze_audio.h",
+ "rtc_event_log_visualizer/analyzer.cc",
+ "rtc_event_log_visualizer/analyzer.h",
+ "rtc_event_log_visualizer/analyzer_common.cc",
+ "rtc_event_log_visualizer/analyzer_common.h",
+ "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",
+ ]
+ deps = [
+ ":chart_proto",
+ "../api:function_view",
+ "../api:network_state_predictor_api",
+ "../modules/audio_coding:neteq_input_audio_tools",
+ "../modules/audio_coding:neteq_tools_minimal",
+ "../rtc_base:ignore_wundef",
+ "../rtc_base:logging",
+ "../rtc_base:macromagic",
+ "../rtc_base:rate_statistics",
+ "../rtc_base:refcount",
+
+ # TODO(kwiberg): Remove this dependency.
+ "../api/audio_codecs:audio_codecs_api",
+ "../api/transport:field_trial_based_config",
+ "../api/transport:goog_cc",
+ "../api/transport:network_control",
+ "../call:call_interfaces",
+ "../call:video_stream_api",
+ "../logging:rtc_event_log_parser",
+ "../logging:rtc_stream_config",
+ "../modules/audio_coding:ana_debug_dump_proto",
+ "../modules/audio_coding:audio_network_adaptor",
+ "../modules/audio_coding:neteq_tools",
+ "../modules/congestion_controller",
+ "../modules/congestion_controller/goog_cc:delay_based_bwe",
+ "../modules/congestion_controller/goog_cc:estimators",
+ "../modules/congestion_controller/rtp:transport_feedback",
+ "../modules/pacing",
+ "../modules/remote_bitrate_estimator",
+ "../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
+ "../rtc_base:rtc_numerics",
+ "../rtc_base:stringutils",
+ "../system_wrappers",
+ "../test:explicit_key_value_config",
+ ]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/algorithm:container",
+ "//third_party/abseil-cpp/absl/base:core_headers",
+ "//third_party/abseil-cpp/absl/functional:bind_front",
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+ }
+ }
+}
+
+if (rtc_include_tests) {
+ if (!build_with_chromium) {
+ if (rtc_enable_protobuf) {
+ rtc_executable("event_log_visualizer") {
+ # TODO(bugs.webrtc.org/14248): Remove once usage of std::tmpnam
+ # is removed (in favor of in memory InputAudioFile.
+ cflags_cc = [ "-Wno-deprecated-declarations" ]
+ sources = [
+ "rtc_event_log_visualizer/conversational_speech_en.h",
+ "rtc_event_log_visualizer/main.cc",
+ ]
+ deps = [
+ ":event_log_visualizer_utils",
+ "../api/neteq:neteq_api",
+ "../api/rtc_event_log",
+ "../logging:rtc_event_log_parser",
+ "../modules/audio_coding:neteq",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "../rtc_base:protobuf_utils",
+ "../system_wrappers:field_trial",
+ "//third_party/abseil-cpp/absl/algorithm:container",
+ "//third_party/abseil-cpp/absl/flags:config",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ "//third_party/abseil-cpp/absl/flags:usage",
+ "//third_party/abseil-cpp/absl/strings",
+ ]
+ }
+
+ rtc_executable("rtc_event_log_to_text") {
+ testonly = true
+ sources = [
+ "rtc_event_log_to_text/converter.cc",
+ "rtc_event_log_to_text/converter.h",
+ "rtc_event_log_to_text/main.cc",
+ ]
+ deps = [
+ "../api/rtc_event_log",
+ "../logging:ice_log",
+ "../logging:rtc_event_audio",
+ "../logging:rtc_event_begin_end",
+ "../logging:rtc_event_bwe",
+ "../logging:rtc_event_frame_events",
+ "../logging:rtc_event_generic_packet_events",
+ "../logging:rtc_event_log2_proto",
+ "../logging:rtc_event_log_impl_encoder",
+ "../logging:rtc_event_log_parser",
+ "../logging:rtc_event_log_proto",
+ "../logging:rtc_event_pacing",
+ "../logging:rtc_event_rtp_rtcp",
+ "../logging:rtc_event_video",
+ "../logging:rtc_stream_config",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "//third_party/abseil-cpp/absl/base:core_headers",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ "//third_party/abseil-cpp/absl/flags:usage",
+ "//third_party/abseil-cpp/absl/strings",
+ ]
+ }
+ }
+
+ tools_unittests_resources = [
+ "../resources/foreman_128x96.yuv",
+ "../resources/foreman_cif.yuv",
+ "../resources/reference_less_video_test_file.y4m",
+ ]
+
+ if (is_ios) {
+ bundle_data("tools_unittests_bundle_data") {
+ testonly = true
+ sources = tools_unittests_resources
+ outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+ }
+ }
+
+ rtc_test("tools_unittests") {
+ testonly = true
+
+ sources = [
+ "frame_analyzer/linear_least_squares_unittest.cc",
+ "frame_analyzer/reference_less_video_analysis_unittest.cc",
+ "frame_analyzer/video_color_aligner_unittest.cc",
+ "frame_analyzer/video_geometry_aligner_unittest.cc",
+ "frame_analyzer/video_quality_analysis_unittest.cc",
+ "frame_analyzer/video_temporal_aligner_unittest.cc",
+ "sanitizers_unittest.cc",
+ "video_file_reader_unittest.cc",
+ "video_file_writer_unittest.cc",
+ ]
+
+ deps = [
+ ":video_file_reader",
+ ":video_file_writer",
+ ":video_quality_analysis",
+ "../api:scoped_refptr",
+ "../api/test/metrics:metric",
+ "../api/test/metrics:metrics_logger",
+ "../api/video:video_frame",
+ "../api/video:video_rtp_headers",
+ "../common_video",
+ "../rtc_base:checks",
+ "../rtc_base:null_socket_server",
+ "../rtc_base:threading",
+ "../system_wrappers",
+ "../test:fileutils",
+ "../test:test_main",
+ "../test:test_support",
+ "//testing/gtest",
+ "//third_party/libyuv",
+ ]
+
+ if (!build_with_chromium) {
+ deps += [ ":reference_less_video_analysis_lib" ]
+ }
+
+ 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) {
+ rtc_executable("audioproc_f") {
+ testonly = true
+ sources = [ "audioproc_f/audioproc_float_main.cc" ]
+ deps = [
+ "../api:audioproc_f_api",
+ "../modules/audio_processing",
+ "../modules/audio_processing:api",
+ ]
+ }
+
+ rtc_executable("unpack_aecdump") {
+ visibility = [ "*" ]
+ sources = [ "unpack_aecdump/unpack.cc" ]
+
+ deps = [
+ "../api:function_view",
+ "../common_audio",
+ "../modules/audio_processing",
+ "../modules/audio_processing:audioproc_debug_proto",
+ "../modules/audio_processing:audioproc_debug_proto",
+ "../modules/audio_processing:audioproc_protobuf_utils",
+ "../rtc_base:checks",
+ "../rtc_base:ignore_wundef",
+ "../rtc_base:macromagic",
+ "../rtc_base:protobuf_utils",
+ "../rtc_base:stringutils",
+ "../rtc_base/system:arch",
+ "//third_party/abseil-cpp/absl/flags:flag",
+ "//third_party/abseil-cpp/absl/flags:parse",
+ ]
+ } # unpack_aecdump
+ }
+ }
+
+ if (rtc_enable_protobuf) {
+ 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",
+ ]
+ outputs = [ "$root_build_dir/{{source_file_part}}" ]
+ deps = [ "../logging:rtc_event_log_proto" ]
+ } # rtp_analyzer
+ }
+}