| # 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_static_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.cc", |
| "channel.h", |
| "channel_proxy.cc", |
| "channel_proxy.h", |
| "conversion.h", |
| "null_audio_poller.cc", |
| "null_audio_poller.h", |
| "remix_resample.cc", |
| "remix_resample.h", |
| "time_interval.cc", |
| "time_interval.h", |
| "transport_feedback_packet_loss_tracker.cc", |
| "transport_feedback_packet_loss_tracker.h", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| |
| deps = [ |
| "..:webrtc_common", |
| "../api:array_view", |
| "../api:call_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:optional", |
| "../api:transport_api", |
| "../api/audio:aec3_factory", |
| "../api/audio:audio_mixer_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../call:bitrate_allocator", |
| "../call:call_interfaces", |
| "../call:rtp_interfaces", |
| "../common_audio", |
| "../common_audio:common_audio_c", |
| "../logging:rtc_event_audio", |
| "../logging:rtc_event_log_api", |
| "../modules:module_api", |
| "../modules/audio_coding", |
| "../modules/audio_coding:audio_format_conversion", |
| "../modules/audio_coding:audio_network_adaptor_config", |
| "../modules/audio_coding:cng", |
| "../modules/audio_device", |
| "../modules/audio_processing", |
| "../modules/bitrate_controller:bitrate_controller", |
| "../modules/pacing:pacing", |
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator", |
| "../modules/rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../modules/utility", |
| "../rtc_base:audio_format_to_string", |
| "../rtc_base:checks", |
| "../rtc_base:rate_limiter", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:safe_minmax", |
| "../rtc_base:stringutils", |
| "../system_wrappers", |
| "../system_wrappers:field_trial_api", |
| "../system_wrappers:metrics_api", |
| "utility:audio_frame_operations", |
| ] |
| } |
| if (rtc_include_tests) { |
| rtc_source_set("audio_end_to_end_test") { |
| testonly = true |
| |
| sources = [ |
| "test/audio_end_to_end_test.cc", |
| "test/audio_end_to_end_test.h", |
| ] |
| deps = [ |
| ":audio", |
| "../system_wrappers:system_wrappers", |
| "../test:test_common", |
| "../test:test_support", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| rtc_source_set("audio_tests") { |
| testonly = true |
| |
| sources = [ |
| "audio_receive_stream_unittest.cc", |
| "audio_send_stream_tests.cc", |
| "audio_send_stream_unittest.cc", |
| "audio_state_unittest.cc", |
| "mock_voe_channel_proxy.h", |
| "remix_resample_unittest.cc", |
| "time_interval_unittest.cc", |
| "transport_feedback_packet_loss_tracker_unittest.cc", |
| ] |
| deps = [ |
| ":audio", |
| ":audio_end_to_end_test", |
| "../api:mock_audio_mixer", |
| "../call:mock_call_interfaces", |
| "../call:mock_rtp_interfaces", |
| "../call:rtp_interfaces", |
| "../call:rtp_receiver", |
| "../common_audio", |
| "../logging:mocks", |
| "../modules:module_api", |
| "../modules/audio_device:mock_audio_device", |
| "../modules/audio_mixer:audio_mixer_impl", |
| "../modules/audio_processing:audio_processing_statistics", |
| "../modules/audio_processing:mocks", |
| "../modules/bitrate_controller:mocks", |
| "../modules/pacing:pacing", |
| "../modules/rtp_rtcp:mock_rtp_rtcp", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_tests_utils", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base:safe_compare", |
| "../system_wrappers:system_wrappers", |
| "../test:audio_codec_mocks", |
| "../test:rtp_test_utils", |
| "../test:test_common", |
| "../test:test_support", |
| "utility:utility_tests", |
| "//testing/gtest", |
| ] |
| |
| if (!rtc_use_memcheck) { |
| # This test is timing dependent, which rules out running on memcheck bots. |
| sources += [ "test/audio_stats_test.cc" ] |
| } |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| if (rtc_enable_protobuf) { |
| rtc_test("low_bandwidth_audio_test") { |
| testonly = true |
| |
| sources = [ |
| "test/low_bandwidth_audio_test.cc", |
| ] |
| |
| deps = [ |
| ":audio_end_to_end_test", |
| "../common_audio", |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers", |
| "../test:fileutils", |
| "../test:test_common", |
| "../test:test_main", |
| "//testing/gtest", |
| ] |
| 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", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163) |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| |
| group("low_bandwidth_audio_perf_test") { |
| testonly = true |
| |
| deps = [ |
| ":low_bandwidth_audio_test", |
| ] |
| |
| data = [ |
| "test/low_bandwidth_audio_test.py", |
| "../resources/voice_engine/audio_tiny16.wav", |
| "../resources/voice_engine/audio_tiny48.wav", |
| ] |
| 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_source_set("audio_perf_tests") { |
| testonly = true |
| |
| sources = [ |
| "test/audio_bwe_integration_test.cc", |
| "test/audio_bwe_integration_test.h", |
| ] |
| deps = [ |
| "../common_audio", |
| "../rtc_base:rtc_base_approved", |
| "../system_wrappers", |
| "../test:field_trial", |
| "../test:fileutils", |
| "../test:single_threaded_task_queue", |
| "../test:test_common", |
| "../test:test_main", |
| "//testing/gtest", |
| ] |
| |
| data = [ |
| "//resources/voice_engine/audio_dtx16.wav", |
| ] |
| |
| if (!build_with_chromium && is_clang) { |
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| } |
| } |
| } |
