# 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_source_set("audio_device") {
  visibility = [ "*" ]
  public_deps = [
    ":audio_device_generic",
  ]
  if (rtc_include_internal_audio_device && is_ios) {
    public_deps += [ ":audio_device_ios_objc" ]
  }
}

if (rtc_include_internal_audio_device && is_ios) {
  rtc_source_set("audio_device_ios_objc") {
    visibility = [ ":audio_device" ]
    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.h",
      "ios/objc/RTCAudioSessionConfiguration.h",
      "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",
    ]
    deps = [
      ":audio_device_generic",
      "../../api:array_view",
      "../../rtc_base:checks",
      "../../rtc_base:gtest_prod",
      "../../rtc_base:rtc_base",
      "../../rtc_base/system:fallthrough",
      "../../sdk:audio_objc",
      "../../sdk:common_objc",
      "../../system_wrappers:metrics_api",
    ]
    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_device_generic") {
  visibility = [
    ":audio_device",
    ":audio_device_ios_objc",
  ]
  public_configs = [ ":audio_device_config" ]

  deps = [
    "..:module_api",
    "../..:webrtc_common",
    "../../:typedefs",
    "../../api:array_view",
    "../../common_audio",
    "../../common_audio:common_audio_c",
    "../../rtc_base:checks",
    "../../rtc_base:deprecation",
    "../../rtc_base:rtc_base_approved",
    "../../rtc_base:rtc_task_queue",
    "../../system_wrappers",
    "../../system_wrappers:metrics_api",
    "../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_default.h",
    "include/audio_device_defines.h",
    "include/fake_audio_device.h",
    "include/test_audio_device.h",
    "include/test_audio_device_impl.cc",
    "include/test_audio_device_impl.h",
  ]

  if (build_with_mozilla) {
    sources += [
      "opensl/single_rw_fifo.cc",
      "opensl/single_rw_fifo.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_audio_device_plays_sinus_tone) {
    defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
  }
  if (rtc_include_internal_audio_device) {
    # TODO(bugs.webrtc.org/8850): remove this when the circular dependency will be fixed.
    check_includes = false
    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_common.h",
        "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 (build_with_mozilla) {
        include_dirs += [
          "/config/external/nspr",
          "/nsprpub/lib/ds",
          "/nsprpub/pr/include",
        ]
      }
    }
    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" ]
        if (rtc_use_x11) {
          libs += [ "X11" ]
          defines += [ "WEBRTC_USE_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_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",
  ]
}

rtc_source_set("mock_audio_device") {
  testonly = true
  sources = [
    "include/mock_audio_device.h",
    "include/mock_audio_transport.h",
    "mock_audio_device_buffer.h",
  ]
  deps = [
    ":audio_device",
    "../../test:test_support",
  ]
}

if (rtc_include_tests) {
  # TODO(kthelgason): Reenable these tests on simulator.
  # See bugs.webrtc.org/7812
  if (is_ios && !use_ios_simulator) {
    rtc_source_set("audio_device_ios_objc_unittests") {
      testonly = true
      visibility = [ ":*" ]
      sources = [
        "ios/audio_device_unittest_ios.mm",
      ]
      deps = [
        ":audio_device",
        ":mock_audio_device",
        "../../rtc_base:rtc_base_approved",
        "../../sdk:audio_objc",
        "../../system_wrappers",
        "../../test:test_support",
        "//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("audio_device_unittests") {
    testonly = true

    sources = [
      "fine_audio_buffer_unittest.cc",
      "include/test_audio_device_impl_unittest.cc",
    ]
    deps = [
      ":audio_device",
      ":mock_audio_device",
      "../../api:array_view",
      "../../api:optional",
      "../../common_audio",
      "../../rtc_base:checks",
      "../../rtc_base:rtc_base_approved",
      "../../system_wrappers",
      "../../test:test_support",
      "../utility:utility",
    ]
    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",
        "../../sdk/android:libjingle_peerconnection_java",
      ]
    }
    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_include_internal_audio_device) {
      defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
    }
  }
}

if (!build_with_chromium && is_android) {
  rtc_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 = [
      "../../rtc_base:base_java",
    ]
  }
}