# Copyright (c) 2014 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("../../build/webrtc.gni") config("audio_device_config") { include_dirs = [ "../include", "include", "dummy", # Contains dummy audio device implementations. ] if (is_ios) { # GN orders flags on a target before flags from configs. In order to be able # suppress the -Wthread-safety-analysis warning, it has come from a config # and can't be on the target directly. # TODO(tkchin): Remove after fixing # https://bugs.chromium.org/p/webrtc/issues/detail?id=5748 cflags = [ "-Wno-thread-safety-analysis" ] } } config("audio_device_warnings_config") { if (is_win && is_clang) { cflags = [ # Disable warnings failing when compiling with Clang on Windows. # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 "-Wno-bool-conversion", "-Wno-delete-non-virtual-dtor", "-Wno-logical-op-parentheses", "-Wno-microsoft-extra-qualification", "-Wno-microsoft-goto", "-Wno-missing-braces", "-Wno-parentheses-equality", "-Wno-reorder", "-Wno-shift-overflow", "-Wno-tautological-compare", # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6265 # for -Wno-thread-safety-analysis "-Wno-thread-safety-analysis", "-Wno-unused-private-field", ] } } rtc_static_library("audio_device") { public_configs = [ ":audio_device_config" ] deps = [ "../..:webrtc_common", "../../base:rtc_base_approved", "../../base:rtc_task_queue", "../../common_audio", "../../system_wrappers", "../utility", ] sources = [ "audio_device_buffer.cc", "audio_device_buffer.h", "audio_device_config.h", "audio_device_generic.cc", "audio_device_generic.h", "dummy/audio_device_dummy.cc", "dummy/audio_device_dummy.h", "dummy/file_audio_device.cc", "dummy/file_audio_device.h", "fine_audio_buffer.cc", "fine_audio_buffer.h", "include/audio_device.h", "include/audio_device_defines.h", ] include_dirs = [] if (is_linux) { include_dirs += [ "linux" ] } if (is_ios) { include_dirs += [ "ios" ] } if (is_mac) { include_dirs += [ "mac" ] } if (is_win) { include_dirs += [ "win" ] } if (is_android) { include_dirs += [ "android" ] } defines = [] cflags = [] if (rtc_include_internal_audio_device) { sources += [ "audio_device_impl.cc", "audio_device_impl.h", ] if (is_android) { sources += [ "android/audio_device_template.h", "android/audio_manager.cc", "android/audio_manager.h", "android/audio_record_jni.cc", "android/audio_record_jni.h", "android/audio_track_jni.cc", "android/audio_track_jni.h", "android/build_info.cc", "android/build_info.h", "android/opensles_common.cc", "android/opensles_common.h", "android/opensles_player.cc", "android/opensles_player.h", "android/opensles_recorder.cc", "android/opensles_recorder.h", ] libs = [ "log", "OpenSLES", ] } if (rtc_use_dummy_audio_file_devices) { defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ] } else { if (is_linux) { sources += [ "linux/alsasymboltable_linux.cc", "linux/alsasymboltable_linux.h", "linux/audio_device_alsa_linux.cc", "linux/audio_device_alsa_linux.h", "linux/audio_mixer_manager_alsa_linux.cc", "linux/audio_mixer_manager_alsa_linux.h", "linux/latebindingsymboltable_linux.cc", "linux/latebindingsymboltable_linux.h", ] defines += [ "LINUX_ALSA" ] libs = [ "dl", "X11", ] if (rtc_include_pulse_audio) { sources += [ "linux/audio_device_pulse_linux.cc", "linux/audio_device_pulse_linux.h", "linux/audio_mixer_manager_pulse_linux.cc", "linux/audio_mixer_manager_pulse_linux.h", "linux/pulseaudiosymboltable_linux.cc", "linux/pulseaudiosymboltable_linux.h", ] defines += [ "LINUX_PULSE" ] } } if (is_mac) { sources += [ "mac/audio_device_mac.cc", "mac/audio_device_mac.h", "mac/audio_mixer_manager_mac.cc", "mac/audio_mixer_manager_mac.h", "mac/portaudio/pa_memorybarrier.h", "mac/portaudio/pa_ringbuffer.c", "mac/portaudio/pa_ringbuffer.h", ] libs = [ # Needed for CoreGraphics: "ApplicationServices.framework", "AudioToolbox.framework", "CoreAudio.framework", # Needed for CGEventSourceKeyState in audio_device_mac.cc: "CoreGraphics.framework", ] } if (is_ios) { public_deps = [ "../../base:rtc_base", "../../sdk:rtc_sdk_common_objc", ] sources += [ "ios/audio_device_ios.h", "ios/audio_device_ios.mm", "ios/audio_device_not_implemented_ios.mm", "ios/audio_session_observer.h", "ios/objc/RTCAudioSession+Configuration.mm", "ios/objc/RTCAudioSession+Private.h", "ios/objc/RTCAudioSession.h", "ios/objc/RTCAudioSession.mm", "ios/objc/RTCAudioSessionConfiguration.h", "ios/objc/RTCAudioSessionConfiguration.m", "ios/objc/RTCAudioSessionDelegateAdapter.h", "ios/objc/RTCAudioSessionDelegateAdapter.mm", "ios/voice_processing_audio_unit.h", "ios/voice_processing_audio_unit.mm", ] configs += [ "//build/config/compiler:enable_arc" ] libs = [ "AudioToolbox.framework", "AVFoundation.framework", "Foundation.framework", "UIKit.framework", ] } if (is_win) { sources += [ "win/audio_device_core_win.cc", "win/audio_device_core_win.h", "win/audio_device_wave_win.cc", "win/audio_device_wave_win.h", "win/audio_mixer_manager_win.cc", "win/audio_mixer_manager_win.h", ] libs = [ # Required for the built-in WASAPI AEC. "dmoguids.lib", "wmcodecdspuuid.lib", "amstrmid.lib", "msdmo.lib", ] } configs += [ ":audio_device_warnings_config" ] } } else { defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ] } if (!build_with_chromium) { sources += [ # Do not link these into Chrome since they contain static data. "dummy/file_audio_device_factory.cc", "dummy/file_audio_device_factory.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" ] } } config("mock_audio_device_config") { if (is_win) { cflags = [ # TODO(phoglund): get rid of 4373 supression when # http://code.google.com/p/webrtc/issues/detail?id=261 is solved. # legacy warning for ignoring const / volatile in signatures. "/wd4373", ] } } if (rtc_include_tests) { rtc_source_set("mock_audio_device") { testonly = true sources = [ "include/mock_audio_device.h", "include/mock_audio_transport.h", ] deps = [ ":audio_device", "../../test:test_support", ] all_dependent_configs = [ ":mock_audio_device_config" ] } } # These tests do not work on ios, see # https://bugs.chromium.org/p/webrtc/issues/detail?id=4755 if (rtc_include_tests && !is_ios) { rtc_executable("audio_device_tests") { testonly = true sources = [ "test/audio_device_test_api.cc", "test/audio_device_test_defines.h", ] deps = [ ":audio_device", "../..:webrtc_common", "../../system_wrappers", "../../test:test_support", "../../test:test_support_main", "../rtp_rtcp", "../utility", "//testing/gtest", ] public_configs = [ ":audio_device_config" ] } }