blob: 7e8f4738f49bed7cf1dc3b7a7ea6649f002a49ac [file] [log] [blame]
# 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:test_video_capturer",
"../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
}
}