# Copyright (c) 2015 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 (is_android) { import("//build/config/android/config.gni") import("//build/config/android/rules.gni") } rtc_library("audio") { sources = [ "audio_level.cc", "audio_level.h", "audio_receive_stream.cc", "audio_receive_stream.h", "audio_send_stream.cc", "audio_send_stream.h", "audio_state.cc", "audio_state.h", "audio_transport_impl.cc", "audio_transport_impl.h", "channel_receive.cc", "channel_receive.h", "channel_receive_frame_transformer_delegate.cc", "channel_receive_frame_transformer_delegate.h", "channel_send.cc", "channel_send.h", "channel_send_frame_transformer_delegate.cc", "channel_send_frame_transformer_delegate.h", "conversion.h", "remix_resample.cc", "remix_resample.h", ] deps = [ "../api:array_view", "../api:call_api", "../api:field_trials_view", "../api:frame_transformer_interface", "../api:function_view", "../api:rtp_headers", "../api:rtp_parameters", "../api:scoped_refptr", "../api:sequence_checker", "../api:transport_api", "../api/audio:aec3_factory", "../api/audio:audio_frame_api", "../api/audio:audio_frame_processor", "../api/audio:audio_mixer_api", "../api/audio_codecs:audio_codecs_api", "../api/crypto:frame_decryptor_interface", "../api/crypto:frame_encryptor_interface", "../api/crypto:options", "../api/neteq:neteq_api", "../api/rtc_event_log", "../api/task_queue", "../api/task_queue:pending_task_safety_flag", "../api/transport/rtp:rtp_source", "../api/units:time_delta", "../call:audio_sender_interface", "../call:bitrate_allocator", "../call:call_interfaces", "../call:rtp_interfaces", "../common_audio", "../common_audio:common_audio_c", "../logging:rtc_event_audio", "../logging:rtc_stream_config", "../media:rtc_media_base", "../modules/async_audio_processing", "../modules/audio_coding", "../modules/audio_coding:audio_coding_module_typedefs", "../modules/audio_coding:audio_encoder_cng", "../modules/audio_coding:audio_network_adaptor_config", "../modules/audio_coding:red", "../modules/audio_device", "../modules/audio_processing", "../modules/audio_processing:api", "../modules/audio_processing:audio_frame_proxies", "../modules/audio_processing:rms_level", "../modules/pacing", "../modules/rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", "../modules/utility:utility", "../rtc_base:audio_format_to_string", "../rtc_base:buffer", "../rtc_base:checks", "../rtc_base:event_tracer", "../rtc_base:logging", "../rtc_base:macromagic", "../rtc_base:race_checker", "../rtc_base:rate_limiter", "../rtc_base:refcount", "../rtc_base:rtc_event", "../rtc_base:rtc_task_queue", "../rtc_base:safe_conversions", "../rtc_base:safe_minmax", "../rtc_base:stringutils", "../rtc_base:threading", "../rtc_base:timeutils", "../rtc_base/containers:flat_set", "../rtc_base/experiments:field_trial_parser", "../rtc_base/synchronization:mutex", "../rtc_base/system:no_unique_address", "../rtc_base/task_utils:repeating_task", "../system_wrappers", "../system_wrappers:field_trial", "../system_wrappers:metrics", "utility:audio_frame_operations", ] absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable", "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/strings", "//third_party/abseil-cpp/absl/types:optional", ] } if (rtc_include_tests) { rtc_library("audio_end_to_end_test") { testonly = true sources = [ "test/audio_end_to_end_test.cc", "test/audio_end_to_end_test.h", ] deps = [ ":audio", "../api:simulated_network_api", "../api/task_queue", "../call:fake_network", "../call:simulated_network", "../system_wrappers", "../test:test_common", "../test:test_support", ] } rtc_library("audio_tests") { testonly = true sources = [ "audio_receive_stream_unittest.cc", "audio_send_stream_tests.cc", "audio_send_stream_unittest.cc", "audio_state_unittest.cc", "channel_receive_frame_transformer_delegate_unittest.cc", "channel_send_frame_transformer_delegate_unittest.cc", "mock_voe_channel_proxy.h", "remix_resample_unittest.cc", "test/audio_stats_test.cc", "test/nack_test.cc", "test/non_sender_rtt_test.cc", ] deps = [ ":audio", ":audio_end_to_end_test", "../api:libjingle_peerconnection_api", "../api:mock_audio_mixer", "../api:mock_frame_decryptor", "../api:mock_frame_encryptor", "../api/audio:audio_frame_api", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs/opus:audio_decoder_opus", "../api/audio_codecs/opus:audio_encoder_opus", "../api/crypto:frame_decryptor_interface", "../api/rtc_event_log", "../api/task_queue:default_task_queue_factory", "../api/task_queue/test:mock_task_queue_base", "../api/units:time_delta", "../call:mock_bitrate_allocator", "../call:mock_call_interfaces", "../call:mock_rtp_interfaces", "../call:rtp_interfaces", "../call:rtp_receiver", "../call:rtp_sender", "../common_audio", "../logging:mocks", "../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", "../modules/audio_mixer:audio_mixer_test_utils", "../modules/audio_processing:audio_processing_statistics", "../modules/audio_processing:mocks", "../modules/pacing", "../modules/rtp_rtcp:mock_rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", "../modules/utility:utility", "../rtc_base:checks", "../rtc_base:macromagic", "../rtc_base:refcount", "../rtc_base:rtc_base_tests_utils", "../rtc_base:safe_compare", "../rtc_base:task_queue_for_test", "../rtc_base:timeutils", "../system_wrappers", "../test:audio_codec_mocks", "../test:field_trial", "../test:mock_frame_transformer", "../test:mock_transformable_frame", "../test:mock_transport", "../test:rtp_test_utils", "../test:scoped_key_value_config", "../test:test_common", "../test:test_support", "../test/time_controller:time_controller", "utility:utility_tests", "//testing/gtest", ] } if (rtc_enable_protobuf && !build_with_chromium) { rtc_test("low_bandwidth_audio_test") { testonly = true sources = [ "test/low_bandwidth_audio_test.cc", "test/low_bandwidth_audio_test_flags.cc", "test/pc_low_bandwidth_audio_test.cc", ] deps = [ ":audio_end_to_end_test", "../api:create_network_emulation_manager", "../api:create_peerconnection_quality_test_fixture", "../api:network_emulation_manager_api", "../api:peer_connection_quality_test_fixture_api", "../api:simulated_network_api", "../api:time_controller", "../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", "../api/test/pclf:media_configuration", "../api/test/pclf:media_quality_test_params", "../api/test/pclf:peer_configurer", "../call:simulated_network", "../common_audio", "../system_wrappers", "../test:fileutils", "../test:test_common", "../test:test_main", "../test:test_support", "../test/pc/e2e:network_quality_metrics_reporter", "//testing/gtest", ] absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag", "//third_party/abseil-cpp/absl/strings", ] if (is_android) { use_default_launcher = false deps += [ "//build/android/gtest_apk:native_test_instrumentation_test_runner_java", "//testing/android/native_test:native_test_java", "//testing/android/native_test:native_test_support", ] } data = [ "../resources/voice_engine/audio_tiny16.wav", "../resources/voice_engine/audio_tiny48.wav", ] } group("low_bandwidth_audio_perf_test") { testonly = true deps = [ ":low_bandwidth_audio_test", "//third_party/catapult/tracing/tracing/proto:histogram_proto", "//third_party/protobuf:py_proto_runtime", ] data = [ "test/low_bandwidth_audio_test.py", "../resources/voice_engine/audio_tiny16.wav", "../resources/voice_engine/audio_tiny48.wav", "${root_out_dir}/pyproto/tracing/tracing/proto/histogram_pb2.py", ] # TODO(http://crbug.com/1029452): Create a cleaner target with just the # tracing python code. We don't need Polymer for instance. data_deps = [ "//third_party/catapult/tracing:convert_chart_json" ] if (is_win) { data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ] } else { data += [ "${root_out_dir}/low_bandwidth_audio_test" ] } if (is_linux || is_chromeos || is_android || is_fuchsia) { data += [ "../tools_webrtc/audio_quality/linux/PolqaOem64", "../tools_webrtc/audio_quality/linux/pesq", ] } if (is_win) { data += [ "../tools_webrtc/audio_quality/win/PolqaOem64.dll", "../tools_webrtc/audio_quality/win/PolqaOem64.exe", "../tools_webrtc/audio_quality/win/pesq.exe", "../tools_webrtc/audio_quality/win/vcomp120.dll", ] } if (is_mac) { data += [ "../tools_webrtc/audio_quality/mac/pesq" ] } } } if (!build_with_chromium) { rtc_library("audio_perf_tests") { testonly = true sources = [ "test/audio_bwe_integration_test.cc", "test/audio_bwe_integration_test.h", ] deps = [ "../api:simulated_network_api", "../api/task_queue", "../call:fake_network", "../call:simulated_network", "../common_audio", "../rtc_base:task_queue_for_test", "../system_wrappers", "../test:field_trial", "../test:fileutils", "../test:test_common", "../test:test_main", "../test:test_support", "//testing/gtest", ] absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ] data = [ "//resources/voice_engine/audio_dtx16.wav" ] } } }