# 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_receive_stream.cc", "audio_receive_stream.h", "audio_send_stream.cc", "audio_send_stream.h", "audio_state.cc", "audio_state.h", "audio_transport_proxy.cc", "audio_transport_proxy.h", "conversion.h", "null_audio_poller.cc", "null_audio_poller.h", "scoped_voe_interface.h", "time_interval.cc", "time_interval.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:audio_mixer_api", "../api:call_api", "../api:optional", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_encoder_factory", "../call:bitrate_allocator", "../call:call_interfaces", "../call:rtp_interfaces", "../common_audio", "../modules/audio_coding:cng", "../modules/audio_device", "../modules/audio_processing", "../modules/bitrate_controller:bitrate_controller", "../modules/congestion_controller:congestion_controller", "../modules/pacing:pacing", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp:rtp_rtcp", "../rtc_base:rtc_base", "../rtc_base:rtc_base_approved", "../rtc_base:rtc_task_queue", "../system_wrappers", "../voice_engine", ] } 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:fake_audio_device", "../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", "time_interval_unittest.cc", ] deps = [ ":audio", ":audio_end_to_end_test", "../api:mock_audio_mixer", "../call:mock_rtp_interfaces", "../call:rtp_interfaces", "../call:rtp_receiver", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", "../modules/audio_processing:audio_processing_statistics", "../modules/congestion_controller:congestion_controller", "../modules/congestion_controller:mock_congestion_controller", "../modules/pacing:mock_paced_sender", "../modules/pacing:pacing", "../modules/rtp_rtcp:mock_rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", "../rtc_base:rtc_base_approved", "../rtc_base:rtc_base_tests_utils", "../rtc_base:rtc_task_queue", "../system_wrappers:system_wrappers", "../test:audio_codec_mocks", "../test:rtp_test_utils", "../test:test_common", "../test:test_support", "../voice_engine", "utility:utility_tests", "//testing/gmock", "//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:fake_audio_device", "../test:test_common", "../test:test_main", "//testing/gmock", "//testing/gtest", ] if (is_android) { deps += [ "//testing/android/native_test:native_test_native_code" ] } data = [ "../resources/voice_engine/audio_dtx16.wav", "../resources/voice_engine/audio_tiny16.wav", "../resources/voice_engine/audio_tiny48.wav", "test/low_bandwidth_audio_test.py", ] if (is_linux) { 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 && 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_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:fake_audio_device", "../test:field_trial", "../test:single_threaded_task_queue", "../test:test_common", "../test:test_main", "//testing/gmock", "//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" ] } } }