Reland of Moving webrtc.gni up one level from build/

webrtc/build/webrtc.gni

Index: webrtc/build/webrtc.gni
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index d179ed4e7dbda3822daa454b986ae409cfe0ef39..04bab4677ecb2c5444b21a86ae261e99770f31fb 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -1,4 +1,4 @@
-# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+# Copyright (c) 2017 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
@@ -6,320 +6,4 @@
 # 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/config/sanitizers/sanitizers.gni")
-import("//build_overrides/build.gni")
-import("//testing/test.gni")
-
-declare_args() {
-  # Disable this to avoid building the Opus audio codec.
-  rtc_include_opus = true
-
-  # Enable this to let the Opus audio codec change complexity on the fly.
-  rtc_opus_variable_complexity = false
-
-  # 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
-
-  # 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
-
-  # Enable when an external authentication mechanism is used for performing
-  # packet authentication for RTP packets instead of libsrtp.
-  rtc_enable_external_auth = build_with_chromium
-
-  # Selects whether debug dumps for the audio processing module
-  # should be generated.
-  apm_debug_dump = false
-
-  # Set this to true to enable BWE test logging.
-  rtc_enable_bwe_test_logging = false
-
-  # Set this to disable building with support for SCTP data channels.
-  rtc_enable_sctp = true
-
-  # 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
-
-  # 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
-
-  # 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 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
-
-  # When set to true, test targets will declare the files needed to run memcheck
-  # as data dependencies. This is to enable memcheck execution on swarming bots.
-  rtc_use_memcheck = 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)
-
-  rtc_restrict_logging = build_with_chromium
-
-  # Excluded in Chromium since its prerequisites don't require Pulse Audio.
-  rtc_include_pulse_audio = !build_with_chromium
-
-  # Chromium uses its own IO handling, so the internal ADM is only built for
-  # standalone WebRTC.
-  rtc_include_internal_audio_device = !build_with_chromium
-
-  # Include tests in standalone checkout.
-  rtc_include_tests = !build_with_chromium
-}
-
-# 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"
-
-# Desktop capturer is supported only on Windows, OSX and Linux.
-rtc_desktop_capture_supported = is_win || is_mac || is_linux
-
-###############################################################################
-# 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")
-
-# Global configuration that should be applied to all WebRTC targets.
-# You normally shouldn't need to include this in your target as it's
-# automatically included when using the rtc_* templates.
-# It sets defines, include paths and compilation warnings accordingly,
-# both for WebRTC stand-alone builds and for the scenario when WebRTC
-# native code is built as part of Chromium.
-rtc_common_configs = [ webrtc_root + ":common_config" ]
-
-# Global public configuration that should be applied to all WebRTC targets. You
-# normally shouldn't need to include this in your target as it's automatically
-# included when using the rtc_* templates. It set the defines, include paths and
-# compilation warnings that should be propagated to dependents of the targets
-# depending on the target having this config.
-rtc_common_inherited_config = webrtc_root + ":common_inherited_config"
-
-# Common configs to remove or add in all rtc targets.
-rtc_remove_configs = []
-rtc_add_configs = rtc_common_configs
-
-set_defaults("rtc_test") {
-  configs = rtc_add_configs
-  suppressed_configs = []
-}
-
-set_defaults("rtc_source_set") {
-  configs = rtc_add_configs
-  suppressed_configs = []
-}
-
-set_defaults("rtc_executable") {
-  configs = rtc_add_configs
-  suppressed_configs = []
-}
-
-set_defaults("rtc_static_library") {
-  configs = rtc_add_configs
-  suppressed_configs = []
-}
-
-set_defaults("rtc_shared_library") {
-  configs = rtc_add_configs
-  suppressed_configs = []
-}
-
-template("rtc_test") {
-  test(target_name) {
-    forward_variables_from(invoker,
-                           "*",
-                           [
-                             "configs",
-                             "public_configs",
-                             "suppressed_configs",
-                           ])
-    configs += invoker.configs
-    configs -= rtc_remove_configs
-    configs -= invoker.suppressed_configs
-    public_configs = [ rtc_common_inherited_config ]
-    if (defined(invoker.public_configs)) {
-      public_configs += invoker.public_configs
-    }
-  }
-}
-
-template("rtc_source_set") {
-  source_set(target_name) {
-    forward_variables_from(invoker,
-                           "*",
-                           [
-                             "configs",
-                             "public_configs",
-                             "suppressed_configs",
-                           ])
-    configs += invoker.configs
-    configs -= rtc_remove_configs
-    configs -= invoker.suppressed_configs
-    public_configs = [ rtc_common_inherited_config ]
-    if (defined(invoker.public_configs)) {
-      public_configs += invoker.public_configs
-    }
-  }
-}
-
-template("rtc_executable") {
-  executable(target_name) {
-    forward_variables_from(invoker,
-                           "*",
-                           [
-                             "deps",
-                             "configs",
-                             "public_configs",
-                             "suppressed_configs",
-                           ])
-    configs += invoker.configs
-    configs -= rtc_remove_configs
-    configs -= invoker.suppressed_configs
-    deps = [
-      "//build/config/sanitizers:deps",
-    ]
-    deps += invoker.deps
-    public_configs = [ rtc_common_inherited_config ]
-    if (defined(invoker.public_configs)) {
-      public_configs += invoker.public_configs
-    }
-  }
-}
-
-template("rtc_static_library") {
-  static_library(target_name) {
-    forward_variables_from(invoker,
-                           "*",
-                           [
-                             "configs",
-                             "public_configs",
-                             "suppressed_configs",
-                           ])
-    configs += invoker.configs
-    configs -= rtc_remove_configs
-    configs -= invoker.suppressed_configs
-    public_configs = [ rtc_common_inherited_config ]
-    if (defined(invoker.public_configs)) {
-      public_configs += invoker.public_configs
-    }
-  }
-}
-
-template("rtc_shared_library") {
-  shared_library(target_name) {
-    forward_variables_from(invoker,
-                           "*",
-                           [
-                             "configs",
-                             "public_configs",
-                             "suppressed_configs",
-                           ])
-    configs += invoker.configs
-    configs -= rtc_remove_configs
-    configs -= invoker.suppressed_configs
-    public_configs = [ rtc_common_inherited_config ]
-    if (defined(invoker.public_configs)) {
-      public_configs += invoker.public_configs
-    }
-  }
-}
+import("../webrtc.gni")