webrtc_m130/webrtc/modules/audio_device/BUILD.gn

# 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("../../webrtc.gni")

if (is_android) {
  import("//build/config/android/config.gni")
  import("//build/config/android/rules.gni")
}

config("audio_device_config") {
  include_dirs = [
    "../include",
    "include",
    "dummy",  # Contains dummy audio device implementations.
  ]
}

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 = [
    "..:module_api",
    "../..: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_data_observer.cc",
      "audio_device_impl.cc",
      "audio_device_impl.h",
      "include/audio_device_data_observer.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",
        ]
        deps += [ ":mac_portaudio" ]
        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:gtest_prod",
          "../../base:rtc_base",
          "../../sdk:objc_common",
        ]
        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",
        ]
        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",
        ]
        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" ]
  }
}

rtc_source_set("mac_portaudio") {
  visibility = [ ":*" ]  # Only targets in this file can depend on this.
  sources = [
    "mac/portaudio/pa_memorybarrier.h",
    "mac/portaudio/pa_ringbuffer.c",
    "mac/portaudio/pa_ringbuffer.h",
  ]
}

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("audio_device_unittests") {
    testonly = true

    # Skip restricting visibility on mobile platforms since the tests on those
    # gets additional generated targets which would require many lines here to
    # cover (which would be confusing to read and hard to maintain).
    if (!is_android && !is_ios) {
      visibility = [ "//webrtc/modules:modules_unittests" ]
    }
    sources = [
      "fine_audio_buffer_unittest.cc",
    ]
    deps = [
      ":audio_device",
      ":mock_audio_device",
      "../../base:rtc_base_approved",
      "../../system_wrappers:system_wrappers",
      "../../test:test_support",
      "../utility:utility",
      "//testing/gmock",
    ]
    if (is_linux || is_mac || is_win) {
      sources += [ "audio_device_unittest.cc" ]
    }
    if (is_android) {
      # Need to disable error due to the line in
      # base/android/jni_android.h triggering it:
      # const BASE_EXPORT jobject GetApplicationContext()
      # error: type qualifiers ignored on function return type
      cflags = [ "-Wno-ignored-qualifiers" ]
      sources += [
        "android/audio_device_unittest.cc",
        "android/audio_manager_unittest.cc",
        "android/ensure_initialized.cc",
        "android/ensure_initialized.h",
      ]
      deps += [
        "../../../base",
        "//webrtc/sdk/android:libjingle_peerconnection_java",
      ]
    }
    if (is_ios) {
      sources += [ "ios/objc/RTCAudioSessionTest.mm" ]
      if (target_cpu != "x64") {
        sources += [ "ios/audio_device_unittest_ios.mm" ]
      }
      deps += [ "//third_party/ocmock" ]
    }
    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("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" ]
  }

  if (!is_ios) {
    # These tests do not work on ios, see
    # https://bugs.chromium.org/p/webrtc/issues/detail?id=4755
    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",
        "../../base:rtc_base_approved",
        "../../system_wrappers",
        "../../test:test_main",
        "../../test:test_support",
        "../rtp_rtcp",
        "../utility",
        "//testing/gtest",
      ]
      public_configs = [ ":audio_device_config" ]
    }
  }
}

if (!build_with_chromium && is_android) {
  android_library("audio_device_java") {
    java_files = [
      "android/java/src/org/webrtc/voiceengine/BuildInfo.java",
      "android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
      "android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
      "android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
      "android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
      "android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
    ]
    deps = [
      "//webrtc/base:base_java",
    ]
  }
}