# Copyright (c) 2016 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/config/linux/pkg_config.gni")
import("../build/webrtc.gni")

group("media") {
  deps = [
    ":rtc_media",
  ]
}

config("rtc_media_defines_config") {
  defines = [
    "HAVE_WEBRTC_VIDEO",
    "HAVE_WEBRTC_VOICE",
  ]
}

config("rtc_media_warnings_config") {
  # GN orders flags on a target before flags from configs. The default config
  # adds these flags so to cancel them out they need to come from a config and
  # cannot be on the target directly.
  if (!is_win) {
    cflags = [ "-Wno-deprecated-declarations" ]
    cflags_cc = [ "-Wno-overloaded-virtual" ]
  }
}

if (is_linux && rtc_use_gtk) {
  pkg_config("gtk-lib") {
    packages = [
      "gobject-2.0",
      "gthread-2.0",
      "gtk+-2.0",
    ]
  }
}

source_set("rtc_media") {
  defines = []
  libs = []
  deps = []
  sources = [
    "base/audiosource.h",
    "base/codec.cc",
    "base/codec.h",
    "base/cpuid.cc",
    "base/cpuid.h",
    "base/cryptoparams.h",
    "base/device.h",
    "base/fakescreencapturerfactory.h",
    "base/hybriddataengine.h",
    "base/mediachannel.h",
    "base/mediacommon.h",
    "base/mediaconstants.cc",
    "base/mediaconstants.h",
    "base/mediaengine.cc",
    "base/mediaengine.h",
    "base/rtpdataengine.cc",
    "base/rtpdataengine.h",
    "base/rtpdump.cc",
    "base/rtpdump.h",
    "base/rtputils.cc",
    "base/rtputils.h",
    "base/screencastid.h",
    "base/streamparams.cc",
    "base/streamparams.h",
    "base/turnutils.cc",
    "base/turnutils.h",
    "base/videoadapter.cc",
    "base/videoadapter.h",
    "base/videobroadcaster.cc",
    "base/videobroadcaster.h",
    "base/videocapturer.cc",
    "base/videocapturer.h",
    "base/videocapturerfactory.h",
    "base/videocommon.cc",
    "base/videocommon.h",
    "base/videoframe.cc",
    "base/videoframe.h",
    "base/videoframefactory.cc",
    "base/videoframefactory.h",
    "base/videorenderer.h",
    "base/videosourcebase.cc",
    "base/videosourcebase.h",
    "base/yuvframegenerator.cc",
    "base/yuvframegenerator.h",
    "devices/videorendererfactory.h",
    "engine/nullwebrtcvideoengine.h",
    "engine/simulcast.cc",
    "engine/simulcast.h",
    "engine/webrtccommon.h",
    "engine/webrtcmediaengine.cc",
    "engine/webrtcmediaengine.h",
    "engine/webrtcvideocapturer.cc",
    "engine/webrtcvideocapturer.h",
    "engine/webrtcvideocapturerfactory.cc",
    "engine/webrtcvideocapturerfactory.h",
    "engine/webrtcvideodecoderfactory.h",
    "engine/webrtcvideoencoderfactory.h",
    "engine/webrtcvideoengine2.cc",
    "engine/webrtcvideoengine2.h",
    "engine/webrtcvideoframe.cc",
    "engine/webrtcvideoframe.h",
    "engine/webrtcvideoframefactory.cc",
    "engine/webrtcvideoframefactory.h",
    "engine/webrtcvoe.h",
    "engine/webrtcvoiceengine.cc",
    "engine/webrtcvoiceengine.h",
    "sctp/sctpdataengine.cc",
    "sctp/sctpdataengine.h",
  ]

  configs += [
    "..:common_config",
    ":rtc_media_warnings_config",
  ]

  public_configs = [ "..:common_inherited_config" ]

  if (is_clang) {
    # Suppress warnings from Chrome's Clang plugins.
    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
    configs -= [ "//build/config/clang:extra_warnings" ]
    configs -= [ "//build/config/clang:find_bad_constructs" ]
  }

  if (is_win) {
    cflags = [
      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
      "/wd4267",  # conversion from "size_t" to "int", possible loss of data.
      "/wd4389",  # signed/unsigned mismatch.
    ]
  }

  if (rtc_build_libyuv) {
    deps += [ "$rtc_libyuv_dir" ]
    public_deps = [
      "$rtc_libyuv_dir",
    ]
  } else {
    # Need to add a directory normally exported by libyuv.
    include_dirs += [ "$rtc_libyuv_dir/include" ]
  }

  if (rtc_build_usrsctp) {
    include_dirs = [
      # TODO(jiayl): move this into the public_configs of
      # //third_party/usrsctp/BUILD.gn.
      "//third_party/usrsctp/usrsctplib",
    ]
    deps += [ "//third_party/usrsctp" ]
  }

  if (build_with_chromium) {
    deps += [ "../modules/video_capture:video_capture" ]
  } else {
    configs += [ ":rtc_media_defines_config" ]
    public_configs += [ ":rtc_media_defines_config" ]
    deps += [ "../modules/video_capture:video_capture_internal_impl" ]
  }
  if (is_linux && rtc_use_gtk) {
    sources += [
      "devices/gtkvideorenderer.cc",
      "devices/gtkvideorenderer.h",
    ]
    public_configs += [ ":gtk-lib" ]
  }
  if (is_win) {
    sources += [
      "devices/gdivideorenderer.cc",
      "devices/gdivideorenderer.h",
    ]
    libs += [
      "d3d9.lib",
      "gdi32.lib",
      "strmiids.lib",
    ]
  }
  if (is_mac && current_cpu == "x86") {
    sources += [
      "devices/carbonvideorenderer.cc",
      "devices/carbonvideorenderer.h",
    ]
    libs += [ "Carbon.framework" ]
  }
  if (is_ios || (is_mac && current_cpu != "x86")) {
    defines += [ "CARBON_DEPRECATED=YES" ]
  }

  deps += [
    "..:webrtc_common",
    "../base:rtc_base_approved",
    "../libjingle/xmllite",
    "../libjingle/xmpp",
    "../p2p",
    "../system_wrappers",
    "../voice_engine",
  ]
}