webrtc/build/webrtc.gni

# 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/config/arm.gni")
import("//build/config/features.gni")
import("//build/config/mips.gni")
import("//build_overrides/webrtc.gni")
import("//testing/test.gni")

declare_args() {
  # Disable this to avoid building the Opus audio codec.
  rtc_include_opus = true

  # Disable to use absolute header paths for some libraries.
  rtc_relative_path = true

  # Used to specify an external Jsoncpp include path when not compiling the
  # library that comes with WebRTC (i.e. rtc_build_json == 0).
  rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"

  # Used to specify an external OpenSSL include path when not compiling the
  # library that comes with WebRTC (i.e. rtc_build_ssl == 0).
  rtc_ssl_root = ""

  # Selects fixed-point code where possible.
  rtc_prefer_fixed_point = false

  # Enable data logging. Produces text files with data logged within engines
  # which can be easily parsed for offline processing.
  rtc_enable_data_logging = false

  # Enables the use of protocol buffers for debug recordings.
  rtc_enable_protobuf = true

  # Disable the code for the intelligibility enhancer by default.
  rtc_enable_intelligibility_enhancer = false

  # Selects whether debug dumps for the audio processing module
  # should be generated.
  apm_debug_dump = false

  # Disable these to not build components which can be externally provided.
  rtc_build_expat = true
  rtc_build_json = true
  rtc_build_libjpeg = true
  rtc_build_libsrtp = true
  rtc_build_libvpx = true
  rtc_libvpx_build_vp9 = true
  rtc_build_libyuv = true
  rtc_build_openmax_dl = true
  rtc_build_opus = true
  rtc_build_ssl = true
  rtc_build_usrsctp = true

  # Disable by default.
  rtc_have_dbus_glib = false

  # Enable to use the Mozilla internal settings.
  build_with_mozilla = false

  rtc_enable_android_opensl = false

  # Link-Time Optimizations.
  # Executes code generation at link-time instead of compile-time.
  # https://gcc.gnu.org/wiki/LinkTimeOptimization
  rtc_use_lto = false

  rtc_restrict_logging = true

  # Set to "func", "block", "edge" for coverage generation.
  # At unit test runtime set UBSAN_OPTIONS="coverage=1".
  # It is recommend to set include_examples=0.
  # Use llvm's sancov -html-report for human readable reports.
  # See http://clang.llvm.org/docs/SanitizerCoverage.html .
  rtc_sanitize_coverage = ""

  # Enable libevent task queues on platforms that support it.
  if (is_win || is_mac || is_ios || is_nacl) {
    rtc_enable_libevent = false
    rtc_build_libevent = false
  } else {
    rtc_enable_libevent = true
    rtc_build_libevent = true
  }

  if (current_cpu == "arm" || current_cpu == "arm64") {
    rtc_prefer_fixed_point = true
  }

  if (!is_ios && (current_cpu != "arm" || arm_version >= 7) &&
      current_cpu != "mips64el") {
    rtc_use_openmax_dl = true
  } else {
    rtc_use_openmax_dl = false
  }

  # Determines whether NEON code will be built.
  rtc_build_with_neon =
      (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"

  # Enable this to use HW H.264 encoder/decoder on iOS PeerConnections.
  # Enabling this may break interop with Android clients that support H264.
  rtc_use_objc_h264 = false

  # Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
  # all platforms except Android and iOS. Because FFmpeg can be built
  # with/without H.264 support, |ffmpeg_branding| has to separately be set to a
  # value that includes H.264, for example "Chrome". If FFmpeg is built without
  # H.264, compilation succeeds but |H264DecoderImpl| fails to initialize. See
  # also: |rtc_initialize_ffmpeg|.
  # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
  # http://www.openh264.org, https://www.ffmpeg.org/
  rtc_use_h264 = proprietary_codecs && !is_android && !is_ios

  # Determines whether QUIC code will be built.
  rtc_use_quic = false

  # By default, use normal platform audio support or dummy audio, but don't
  # use file-based audio playout and record.
  rtc_use_dummy_audio_file_devices = false

  # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
  # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
  # only be initialized once. Projects that initialize FFmpeg externally, such
  # as Chromium, must turn this flag off so that WebRTC does not also
  # initialize.
  rtc_initialize_ffmpeg = !build_with_chromium

  # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
  # build environments, even if available for Chromium builds.
  rtc_use_gtk = !build_with_chromium
}

# A second declare_args block, so that declarations within it can
# depend on the possibly overridden variables in the first
# declare_args block.
declare_args() {
  # Include the iLBC audio codec?
  rtc_include_ilbc = !(build_with_chromium || build_with_mozilla)
}

# Make it possible to provide custom locations for some libraries (move these
# up into declare_args should we need to actually use them for the GN build).
rtc_libvpx_dir = "//third_party/libvpx"
rtc_libyuv_dir = "//third_party/libyuv"
rtc_opus_dir = "//third_party/opus"

###############################################################################
# Templates
#

# Points to //webrtc/ in webrtc stand-alone or to //third_party/webrtc/ in
# chromium.
# We need absolute paths for all configs in templates as they are shared in
# different subdirectories.
webrtc_root = get_path_info("../", "abspath")

# Common configs to remove or add in all rtc targets.
rtc_remove_configs = []
rtc_add_configs = []

set_defaults("rtc_test") {
  configs = []
  suppressed_configs = []
}

set_defaults("rtc_source_set") {
  configs = []
  suppressed_configs = []
}

set_defaults("rtc_executable") {
  configs = []
  suppressed_configs = []
}

set_defaults("rtc_static_library") {
  configs = []
  suppressed_configs = []
}

template("rtc_test") {
  test(target_name) {
    forward_variables_from(invoker,
                           "*",
                           [
                             "configs",
                             "suppressed_configs",
                           ])
    configs += invoker.configs
    configs += rtc_add_configs
    configs -= rtc_remove_configs
    configs -= invoker.suppressed_configs
  }
}

template("rtc_source_set") {
  source_set(target_name) {
    forward_variables_from(invoker,
                           "*",
                           [
                             "configs",
                             "suppressed_configs",
                           ])
    configs += invoker.configs
    configs += rtc_add_configs
    configs -= rtc_remove_configs
    configs -= invoker.suppressed_configs
  }
}

template("rtc_executable") {
  executable(target_name) {
    forward_variables_from(invoker,
                           "*",
                           [
                             "configs",
                             "suppressed_configs",
                           ])
    configs += invoker.configs
    configs += rtc_add_configs
    configs -= rtc_remove_configs
    configs -= invoker.suppressed_configs
  }
}

template("rtc_static_library") {
  static_library(target_name) {
    forward_variables_from(invoker,
                           "*",
                           [
                             "configs",
                             "suppressed_configs",
                           ])
    configs += invoker.configs
    configs += rtc_add_configs
    configs -= rtc_remove_configs
    configs -= invoker.suppressed_configs
  }
}