| # 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", |
| "null_audio_poller.cc", |
| "null_audio_poller.h", |
| "remix_resample.cc", |
| "remix_resample.h", |
| ] |
| |
| deps = [ |
| "../api:array_view", |
| "../api:call_api", |
| "../api:frame_transformer_interface", |
| "../api:function_view", |
| "../api:rtp_headers", |
| "../api:rtp_parameters", |
| "../api:scoped_refptr", |
| "../api:transport_api", |
| "../api/audio:aec3_factory", |
| "../api/audio:audio_frame_api", |
| "../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/transport/rtp:rtp_source", |
| "../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", |
| "../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/remote_bitrate_estimator", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/utility", |
| "../rtc_base", |
| "../rtc_base:audio_format_to_string", |
| "../rtc_base:checks", |
| "../rtc_base:rate_limiter", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:safe_minmax", |
| "../rtc_base/experiments:field_trial_parser", |
| "../rtc_base/synchronization:mutex", |
| "../rtc_base/synchronization:sequence_checker", |
| "../rtc_base/task_utils:to_queued_task", |
| "../system_wrappers", |
| "../system_wrappers:field_trial", |
| "../system_wrappers:metrics", |
| "utility:audio_frame_operations", |
| ] |
| absl_deps = [ |
| "//third_party/abseil-cpp/absl/memory", |
| "//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", |
| ] |
| 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/rtc_event_log", |
| "../api/task_queue:default_task_queue_factory", |
| "../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", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../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:test_common", |
| "../test:test_support", |
| "utility:utility_tests", |
| "//testing/gtest", |
| ] |
| } |
| |
| if (rtc_enable_protobuf) { |
| 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", |
| "../call:simulated_network", |
| "../common_audio", |
| "../system_wrappers", |
| "../test:fileutils", |
| "../test:perf_test", |
| "../test:test_common", |
| "../test:test_main", |
| "../test:test_support", |
| "../test/pc/e2e:network_quality_metrics_reporter", |
| "//testing/gtest", |
| "//third_party/abseil-cpp/absl/flags:flag", |
| ] |
| if (is_android) { |
| deps += [ "//testing/android/native_test:native_test_native_code" ] |
| } |
| 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 += [ "${root_out_dir}/low_bandwidth_audio_test" ] |
| } |
| |
| if (is_linux || is_android) { |
| data += [ |
| "../tools_webrtc/audio_quality/linux/PolqaOem64", |
| "../tools_webrtc/audio_quality/linux/pesq", |
| ] |
| } |
| if (is_win) { |
| data += [ |
| "../tools_webrtc/audio_quality/win/PolqaOem64.dll", |
| "../tools_webrtc/audio_quality/win/PolqaOem64.exe", |
| "../tools_webrtc/audio_quality/win/pesq.exe", |
| "../tools_webrtc/audio_quality/win/vcomp120.dll", |
| ] |
| } |
| if (is_mac) { |
| data += [ "../tools_webrtc/audio_quality/mac/pesq" ] |
| } |
| |
| write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps" |
| } |
| } |
| |
| 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:rtc_base_approved", |
| "../rtc_base:task_queue_for_test", |
| "../system_wrappers", |
| "../test:field_trial", |
| "../test:fileutils", |
| "../test:test_common", |
| "../test:test_main", |
| "../test:test_support", |
| "//testing/gtest", |
| ] |
| |
| data = [ "//resources/voice_engine/audio_dtx16.wav" ] |
| } |
| } |
