# Copyright 2014 The Chromium Authors. All rights reserved.
# Use of this source code is governed by a BSD-style license that can be
# found in the LICENSE file.

import("//tools/grit/grit_rule.gni")

# Applied by targets internal to content.
config("content_implementation") {
  defines = [ "CONTENT_IMPLEMENTATION" ]
}

content_shared_components = [
  "//content/gpu",
  "//content/plugin",
  "//content/ppapi_plugin",
  "//content/public/app",
  "//content/public/browser:sources",
  "//content/public/child",
  "//content/public/common",
  "//content/public/plugin",
  "//content/public/renderer",
  "//content/renderer",
  "//content/utility",
]

if (is_component_build) {
  shared_library("content") {
    deps = content_shared_components + [
      "//content/app",
    ]
    forward_dependent_configs_from = deps
  }
} else {
  group("content") {
    deps = content_shared_components
  }
}

grit("resources") {
  source = "content_resources.grd"
  outputs = [
    "grit/content_resources.h",
    "content_resources.pak",
    "content_resources.rc",
  ]
}

# This target exists to "hold" the content_export header so we can do proper
# inclusion testing of it.
source_set("export") {
  visibility = "//content/*"
  sources = [
    "content/common/content_export.h"
  ]
}

# Stubs ------------------------------------------------------------------------

# TODO(brettw) remove this and add a proper dependency on libjingle once that
# target has been converted to GN. This config sets up the include directories
# so content can compile in the meantime.
config("libjingle_stub_config") {
  include_dirs = [
    "//third_party/libjingle/overrides",
    "//third_party/libjingle/source",
    "//third_party/libyuv/include",
    "//third_party/usrsctp",


    "//third_party/webrtc/overrides",  # Must be before webrtc abd third_party.
    "//third_party/webrtc",
    "//third_party",
  ]

  defines = [
    "FEATURE_ENABLE_SSL",
    "FEATURE_ENABLE_VOICEMAIL",
    "EXPAT_RELATIVE_PATH",
    "GTEST_RELATIVE_PATH",
    "NO_MAIN_THREAD_WRAPPING",
    "NO_SOUND_SYSTEM",
  ]

  if (is_mac) {
    defines += [ "OSX", "WEBRTC_MAC" ]
  } else if (is_linux) {
    defines += [ "LINUX", "WEBRTC_LINUX" ]
  } else if (is_android) {
    defines += [ "ANDROID", "WEBRTC_LINUX", "WEBRTC_ANDROID" ]
  } else if (is_win) {
    libs = [ "secur32.lib", "crypt32.lib", "iphlpapi.lib" ]
    defines += [ "WEBRTC_WIN" ]
  }

  if (is_posix) {
    defines += [ "POSIX", "WEBRTC_POSIX" ]
  }
  if (is_chromeos) {
    defines += [ "CHROMEOS" ]
  }
}

# Sets up include dirs while webrtc is being converted to GN.
# TODO(GYP) remove when webrtc is ready.
config("webrtc_stub_config") {
  defines = [ "WEBRTC_CHROMIUM_BUILD" ]
  include_dirs = [
    "//third_party/webrtc/overrides",  # Must be first.
    "//third_party",
  ]

  if (is_mac) {
    defines += [ "WEBRTC_MAC" ]
  } else if (is_win) {
    defines += [ "WEBRTC_WIN" ]
  } else if (is_linux) {
    defines += [ "WEBRTC_LINUX" ]
  } else if (is_ios) {
    defines += [ "WEBRTC_MAC", "WEBRTC_IOS" ]
  } else if (is_android) {
    defines += [ "WEBRTC_LINUX", "WEBRTC_ANDROID" ]
    import("//build/config/crypto.gni")
    if (use_openssl) {
      defines += [ "WEBRTC_ANDROID_OPENSLEQS" ]
    }
  }

  if (is_posix) {
    defines += [ "WEBRTC_POSIX" ]
  }
}