mirror of
https://github.com/mollyim/webrtc.git
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691 lines
24 KiB
Text
691 lines
24 KiB
Text
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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if (rtc_enable_protobuf) {
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import("//third_party/protobuf/proto_library.gni")
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}
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group("rtc_tools") {
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# This target shall build all targets in tools/.
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testonly = true
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deps = [ ":video_file_reader" ]
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if (!build_with_chromium) {
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deps += [
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":frame_analyzer",
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":psnr_ssim_analyzer",
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":video_quality_analysis",
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]
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}
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if (!build_with_chromium && rtc_enable_protobuf) {
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deps += [ ":chart_proto" ]
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}
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if (!build_with_chromium && rtc_include_tests) {
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deps += [ ":tools_unittests" ]
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}
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if (rtc_include_tests && rtc_enable_protobuf) {
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deps += [
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":rtp_analyzer",
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"network_tester",
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]
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}
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if (rtc_include_tests && rtc_enable_protobuf && !build_with_chromium) {
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deps += [
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":audioproc_f",
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":event_log_visualizer",
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":rtc_event_log_to_text",
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":unpack_aecdump",
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]
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}
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if (!build_with_chromium && rtc_enable_grpc) {
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deps += [ "data_channel_benchmark" ]
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}
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}
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rtc_library("video_file_reader") {
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sources = [
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"video_file_reader.cc",
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"video_file_reader.h",
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]
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deps = [
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"../api:make_ref_counted",
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"../api:ref_count",
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"../api:scoped_refptr",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../rtc_base:checks",
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"../rtc_base:logging",
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"../rtc_base:refcount",
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"../rtc_base:stringutils",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("video_file_writer") {
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sources = [
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"video_file_writer.cc",
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"video_file_writer.h",
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]
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deps = [
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":video_file_reader",
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"../api:scoped_refptr",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../rtc_base:logging",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("video_quality_analysis") {
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testonly = true
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sources = [
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"frame_analyzer/linear_least_squares.cc",
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"frame_analyzer/linear_least_squares.h",
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"frame_analyzer/video_color_aligner.cc",
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"frame_analyzer/video_color_aligner.h",
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"frame_analyzer/video_geometry_aligner.cc",
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"frame_analyzer/video_geometry_aligner.h",
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"frame_analyzer/video_quality_analysis.cc",
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"frame_analyzer/video_quality_analysis.h",
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"frame_analyzer/video_temporal_aligner.cc",
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"frame_analyzer/video_temporal_aligner.h",
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]
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deps = [
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":video_file_reader",
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"../api:array_view",
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"../api:make_ref_counted",
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"../api:scoped_refptr",
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"../api/numerics",
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"../api/test/metrics:metric",
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"../api/test/metrics:metrics_logger",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../common_video",
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"../rtc_base:checks",
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"../rtc_base:logging",
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"//third_party/libyuv",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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# TODO(bugs.webrtc.org/11474): Enable this on win if needed. For now it
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# is only required for Linux and Android.
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if (!build_with_chromium && !build_with_mozilla && !is_win && !is_ios) {
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action("frame_analyzer_host") {
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script = "//tools_webrtc/executable_host_build.py"
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outputs = [ "${root_out_dir}/frame_analyzer_host" ]
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args = [
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"--executable_name",
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"frame_analyzer",
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]
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}
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}
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if (!is_component_build) {
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# This target can be built from Chromium but it doesn't support
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# is_component_build=true because it depends on WebRTC testonly code
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# which is not part of //third_party/webrtc_overrides:webrtc_component.
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# Abseil dependencies are not moved to the absl_deps field deliberately.
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# If build_with_chromium is true, the absl_deps replaces the dependencies with
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# the "//third_party/abseil-cpp:absl" target. Which doesn't include absl/flags
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# (and some others) because they cannot be used in Chromiums. Special exception
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# for the "frame_analyzer" target in "third_party/abseil-cpp/absl.gni" allows
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# it to be build in chromium.
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rtc_executable("frame_analyzer") {
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visibility = [ "*" ]
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testonly = true
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sources = [ "frame_analyzer/frame_analyzer.cc" ]
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deps = [
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":video_file_reader",
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":video_file_writer",
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":video_quality_analysis",
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"../api:make_ref_counted",
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"../api:scoped_refptr",
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"../api/test/metrics:chrome_perf_dashboard_metrics_exporter",
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"../api/test/metrics:global_metrics_logger_and_exporter",
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"../api/test/metrics:metrics_exporter",
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"../api/test/metrics:stdout_metrics_exporter",
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"../rtc_base:stringutils",
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"//third_party/abseil-cpp/absl/flags:flag",
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"//third_party/abseil-cpp/absl/flags:parse",
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"//third_party/abseil-cpp/absl/strings",
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]
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if (build_with_chromium) {
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# When building from Chromium, WebRTC's metrics and field trial
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# implementations need to be replaced by the Chromium ones.
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deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
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}
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}
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# This target can be built from Chromium but it doesn't support
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# is_component_build=true because it depends on WebRTC testonly code
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# which is not part of //third_party/webrtc_overrides:webrtc_component.
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# Abseil dependencies are not moved to the absl_deps field deliberately.
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# If build_with_chromium is true, the absl_deps replaces the dependencies with
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# the "//third_party/abseil-cpp:absl" target. Which doesn't include absl/flags
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# (and some others) because they cannot be used in Chromiums. Special exception
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# for the "rtp_generator" target in "third_party/abseil-cpp/absl.gni" allows
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# it to be build in chromium.
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rtc_executable("rtp_generator") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"rtp_generator/main.cc",
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"rtp_generator/rtp_generator.cc",
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"rtp_generator/rtp_generator.h",
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]
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deps = [
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# RingRTC: Transitive dep on opus encode/decode via audio_coding:red.
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"..//api/audio_codecs:builtin_audio_decoder_factory",
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"..//api/audio_codecs:builtin_audio_encoder_factory",
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"..//api/video_codecs:video_decoder_factory_template",
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"..//api/video_codecs:video_decoder_factory_template_dav1d_adapter",
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"..//api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
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"..//api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
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"..//api/video_codecs:video_decoder_factory_template_open_h264_adapter",
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"..//api/video_codecs:video_encoder_factory_template",
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"..//api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
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"..//api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
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"..//api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
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"..//api/video_codecs:video_encoder_factory_template_open_h264_adapter",
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"..//test/network:simulated_network",
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"../api:create_frame_generator",
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"../api:rtp_parameters",
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"../api:transport_api",
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"../api/environment",
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"../api/environment:environment_factory",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../api/video_codecs:video_codecs_api",
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"../call",
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"../call:call_interfaces",
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"../call:fake_network",
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"../call:rtp_interfaces",
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"../call:rtp_sender",
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"../call:simulated_packet_receiver",
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"../call:video_stream_api",
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"../media:media_constants",
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"../media:rtc_audio_video",
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"../rtc_base:rtc_json",
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"../rtc_base:threading",
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"../rtc_base/system:file_wrapper",
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"../test:fileutils",
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"../test:frame_generator_capturer",
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"../test:rtp_test_utils",
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"../video/config:encoder_config",
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"../video/config:streams_config",
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"//third_party/abseil-cpp/absl/flags:flag",
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"//third_party/abseil-cpp/absl/flags:parse",
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"//third_party/abseil-cpp/absl/flags:usage",
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"//third_party/abseil-cpp/absl/strings",
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]
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if (build_with_chromium) {
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# When building from Chromium, WebRTC's metrics and field trial
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# implementations need to be replaced by the Chromium ones.
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deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
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}
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}
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# This target can be built from Chromium but it doesn't support
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# is_component_build=true because it depends on WebRTC testonly code
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# which is not part of //third_party/webrtc_overrides:webrtc_component.
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# Abseil dependencies are not moved to the absl_deps field deliberately.
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# If build_with_chromium is true, the absl_deps replaces the dependencies with
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# the "//third_party/abseil-cpp:absl" target. Which doesn't include absl/flags
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# (and some others) because they cannot be used in Chromiums. Special exception
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# for the "video_replay" target in "third_party/abseil-cpp/absl.gni" allows
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# it to be build in chromium.
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rtc_executable("video_replay") {
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visibility = [ "*" ]
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testonly = true
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sources = [ "video_replay.cc" ]
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deps = [
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"../api:field_trials",
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"../api:rtp_parameters",
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# RingRTC: Transitive dep on opus encode/decode via audio_coding:red.
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/environment",
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"../api/environment:environment_factory",
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"../api/task_queue",
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"../api/test/video:function_video_factory",
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"../api/transport:field_trial_based_config",
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"../api/units:timestamp",
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"../api/video:video_frame",
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"../api/video_codecs:video_codecs_api",
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"../call",
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"../call:call_interfaces",
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"../common_video",
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"../media:rtc_internal_video_codecs",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/video_coding:video_coding_utility",
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"../rtc_base:checks",
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"../rtc_base:rtc_json",
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"../rtc_base:stringutils",
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"../system_wrappers",
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"../test:call_config_utils",
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"../test:encoder_settings",
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"../test:fake_video_codecs",
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"../test:null_transport",
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"../test:rtp_test_utils",
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"../test:run_loop",
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"../test:run_test",
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"../test:run_test_interface",
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"../test:test_renderer",
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"../test:test_support",
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"../test:test_video_capturer",
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"../test:video_test_common",
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"../test:video_test_constants",
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"../test:video_test_support",
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"../test/time_controller:time_controller",
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"//third_party/abseil-cpp/absl/flags:flag",
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"//third_party/abseil-cpp/absl/flags:parse",
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"//third_party/abseil-cpp/absl/strings:strings",
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]
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if (build_with_chromium) {
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# When building from Chromium, WebRTC's metrics and field trial
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# implementations need to be replaced by the Chromium ones.
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deps += [ "//third_party/webrtc_overrides:webrtc_component" ]
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}
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}
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}
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# Only expose the targets needed by Chromium (e.g. frame_analyzer) to avoid
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# building a lot of redundant code as part of Chromium builds.
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if (!build_with_chromium) {
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rtc_executable("psnr_ssim_analyzer") {
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testonly = true
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sources = [ "psnr_ssim_analyzer/psnr_ssim_analyzer.cc" ]
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deps = [
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":video_file_reader",
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":video_quality_analysis",
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"../api:scoped_refptr",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"//third_party/abseil-cpp/absl/flags:flag",
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"//third_party/abseil-cpp/absl/flags:parse",
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"//third_party/abseil-cpp/absl/flags:usage",
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]
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}
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rtc_library("reference_less_video_analysis_lib") {
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testonly = true
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sources = [
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"frame_analyzer/reference_less_video_analysis_lib.cc",
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"frame_analyzer/reference_less_video_analysis_lib.h",
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]
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deps = [
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":video_file_reader",
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":video_quality_analysis",
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"../api:scoped_refptr",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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]
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}
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rtc_executable("reference_less_video_analysis") {
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testonly = true
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sources = [ "frame_analyzer/reference_less_video_analysis.cc" ]
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deps = [
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":reference_less_video_analysis_lib",
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"//third_party/abseil-cpp/absl/flags:flag",
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"//third_party/abseil-cpp/absl/flags:parse",
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"//third_party/abseil-cpp/absl/flags:usage",
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]
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}
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if (rtc_enable_protobuf) {
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proto_library("chart_proto") {
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visibility = [ "*" ]
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sources = [
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"rtc_event_log_visualizer/proto/chart.proto",
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"rtc_event_log_visualizer/proto/chart_enums.proto",
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]
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proto_out_dir = "rtc_tools/rtc_event_log_visualizer/proto"
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}
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rtc_library("event_log_visualizer_utils") {
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visibility = [ "*" ]
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allow_poison = [ "environment_construction" ]
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sources = [
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"rtc_event_log_visualizer/alerts.cc",
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"rtc_event_log_visualizer/alerts.h",
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"rtc_event_log_visualizer/analyze_audio.cc",
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"rtc_event_log_visualizer/analyze_audio.h",
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"rtc_event_log_visualizer/analyzer.cc",
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"rtc_event_log_visualizer/analyzer.h",
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"rtc_event_log_visualizer/analyzer_common.cc",
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"rtc_event_log_visualizer/analyzer_common.h",
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"rtc_event_log_visualizer/log_simulation.cc",
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"rtc_event_log_visualizer/log_simulation.h",
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"rtc_event_log_visualizer/plot_base.cc",
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"rtc_event_log_visualizer/plot_base.h",
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]
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deps = [
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":chart_proto",
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"../api:dtls_transport_interface",
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"../api:function_view",
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"../api:make_ref_counted",
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"../api:network_state_predictor_api",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api/audio_codecs:audio_codecs_api", # TODO(kwiberg): Remove this
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# dependency.
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"../api/environment:environment_factory",
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"../api/neteq:neteq_api",
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"../api/rtc_event_log:rtc_event_log",
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"../api/transport:field_trial_based_config",
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"../api/transport:goog_cc",
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"../api/transport:network_control",
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"../api/units:data_rate",
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"../api/units:time_delta",
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"../api/units:timestamp",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../logging:ice_log",
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"../logging:rtc_event_audio",
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"../logging:rtc_event_audio",
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"../logging:rtc_event_bwe",
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"../logging:rtc_event_generic_packet_events",
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"../logging:rtc_event_log_parser",
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"../logging:rtc_event_rtp_rtcp",
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"../logging:rtc_stream_config",
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"../modules/audio_coding:ana_debug_dump_proto",
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"../modules/audio_coding:audio_network_adaptor",
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"../modules/audio_coding:neteq_input_audio_tools",
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"../modules/audio_coding:neteq_tools",
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"../modules/audio_coding:neteq_tools_minimal",
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"../modules/congestion_controller",
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"../modules/congestion_controller/goog_cc:delay_based_bwe",
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"../modules/congestion_controller/goog_cc:estimators",
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"../modules/congestion_controller/rtp:transport_feedback",
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"../modules/pacing",
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"../modules/remote_bitrate_estimator",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:checks",
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"../rtc_base:logging",
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"../rtc_base:macromagic",
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"../rtc_base:rate_statistics",
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"../rtc_base:refcount",
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"../rtc_base:rtc_numerics",
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"../rtc_base:stringutils",
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"../rtc_base/network:sent_packet",
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"../system_wrappers",
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"../test:explicit_key_value_config",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//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",
|
|
]
|
|
}
|
|
|
|
rtc_library("event_log_visualizer_bindings") {
|
|
visibility = [ "*" ]
|
|
allow_poison = [ "environment_construction" ]
|
|
sources = [
|
|
"rtc_event_log_visualizer/analyzer_bindings.cc",
|
|
"rtc_event_log_visualizer/analyzer_bindings.h",
|
|
]
|
|
deps = [
|
|
":chart_proto",
|
|
":event_log_visualizer_utils",
|
|
"//api/units:time_delta",
|
|
"//logging:rtc_event_log_parser",
|
|
"//rtc_base:protobuf_utils",
|
|
"//rtc_base:safe_conversions",
|
|
]
|
|
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
|
|
}
|
|
|
|
rtc_library("event_log_visualizer_bindings_unittest") {
|
|
testonly = true
|
|
sources = [ "rtc_event_log_visualizer/analyzer_bindings_unittest.cc" ]
|
|
deps = [
|
|
":chart_proto",
|
|
":event_log_visualizer_bindings",
|
|
"//rtc_base:protobuf_utils",
|
|
"//rtc_base/system:file_wrapper",
|
|
"//test:fileutils",
|
|
"//test:test_support",
|
|
]
|
|
}
|
|
}
|
|
|
|
rtc_executable("video_encoder") {
|
|
visibility = [ "*" ]
|
|
testonly = true
|
|
sources = [
|
|
"video_encoder/encoded_image_file_writer.cc",
|
|
"video_encoder/encoded_image_file_writer.h",
|
|
"video_encoder/video_encoder.cc",
|
|
]
|
|
deps = [
|
|
"//api:create_frame_generator",
|
|
"//api:frame_generator_api",
|
|
"//api/environment",
|
|
"//api/environment:environment_factory",
|
|
"//api/video:builtin_video_bitrate_allocator_factory",
|
|
"//api/video_codecs:builtin_video_encoder_factory",
|
|
"//api/video_codecs:video_codecs_api",
|
|
"//media:media_constants",
|
|
"//modules/video_coding:video_codec_interface",
|
|
"//modules/video_coding:video_coding_utility",
|
|
"//modules/video_coding/codecs/av1:av1_svc_config",
|
|
"//modules/video_coding/svc:scalability_mode_util",
|
|
"//rtc_base:logging",
|
|
"//test:y4m_frame_generator",
|
|
"//third_party/abseil-cpp/absl/flags:flag",
|
|
"//third_party/abseil-cpp/absl/flags:parse",
|
|
"//third_party/abseil-cpp/absl/flags:usage",
|
|
]
|
|
}
|
|
}
|
|
|
|
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",
|
|
"../resources/rtc_event_log/rtc_event_log_500kbps.binarypb",
|
|
]
|
|
|
|
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 += [
|
|
":event_log_visualizer_bindings_unittest",
|
|
"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",
|
|
"../api/audio:audio_processing",
|
|
"../modules/audio_processing",
|
|
]
|
|
}
|
|
|
|
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: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
|
|
}
|
|
}
|