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Diffstat (limited to 'third_party/libwebrtc/audio/BUILD.gn')
| -rw-r--r-- | third_party/libwebrtc/audio/BUILD.gn | 329 |
1 files changed, 329 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/BUILD.gn b/third_party/libwebrtc/audio/BUILD.gn new file mode 100644 index 0000000000..4a904aaf28 --- /dev/null +++ b/third_party/libwebrtc/audio/BUILD.gn @@ -0,0 +1,329 @@ +# 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", + "../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", + "../rtc_base:audio_format_to_string", + "../rtc_base:buffer", + "../rtc_base:checks", + "../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/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", + "../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", + "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", + "../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", + ] + 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) { + 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" ] + } + } +} |