7c112f3e5a
This is to allow not building Opus. On non-chromium non-gyp chases, one can let WebRTC depend on other Opus builds. BUG= R=kjellander@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43739004 Cr-Commit-Position: refs/heads/master@{#8754} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8754 4adac7df-926f-26a2-2b94-8c16560cd09d
117 lines
3.8 KiB
Plaintext
117 lines
3.8 KiB
Plaintext
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/config/arm.gni")
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import("//build/config/mips.gni")
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declare_args() {
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# Assume Chromium build for now, since that's the priority case for getting GN
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# up and running with WebRTC.
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build_with_chromium = true
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build_with_libjingle = true
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# Disable this to avoid building the Opus audio codec.
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rtc_include_opus = true
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# Used to specify an external Jsoncpp include path when not compiling the
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# library that comes with WebRTC (i.e. rtc_build_json == 0).
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rtc_jsoncpp_root = "//third_party/jsoncpp/source/include"
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# Used to specify an external OpenSSL include path when not compiling the
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# library that comes with WebRTC (i.e. rtc_build_ssl == 0).
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rtc_ssl_root = ""
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# Selects fixed-point code where possible.
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rtc_prefer_fixed_point = false
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# Enable data logging. Produces text files with data logged within engines
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# which can be easily parsed for offline processing.
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rtc_enable_data_logging = false
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# Enables the use of protocol buffers for debug recordings.
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rtc_enable_protobuf = true
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# Disable these to not build components which can be externally provided.
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rtc_build_json = true
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rtc_build_libjpeg = true
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rtc_build_libyuv = true
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rtc_build_libvpx = true
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rtc_build_vp9 = true
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rtc_build_ssl = true
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rtc_build_opus = true
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# Disable by default.
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rtc_have_dbus_glib = false
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# Enable to use the Mozilla internal settings.
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build_with_mozilla = false
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rtc_enable_android_opensl = false
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# Link-Time Optimizations.
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# Executes code generation at link-time instead of compile-time.
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# https://gcc.gnu.org/wiki/LinkTimeOptimization
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rtc_use_lto = false
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if (build_with_chromium) {
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# Exclude pulse audio on Chromium since its prerequisites don't require
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# pulse audio.
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rtc_include_pulse_audio = false
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# Exclude internal ADM since Chromium uses its own IO handling.
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rtc_include_internal_audio_device = false
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} else {
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# Settings for the standalone (not-in-Chromium) build.
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# TODO(andrew): For now, disable the Chrome plugins, which causes a
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# flood of chromium-style warnings. Investigate enabling them:
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# http://code.google.com/p/webrtc/issues/detail?id=163
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clang_use_chrome_plugins = false
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rtc_include_pulse_audio = true
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rtc_include_internal_audio_device = true
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}
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if (build_with_libjingle) {
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rtc_include_tests = false
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rtc_restrict_logging = true
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} else {
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rtc_include_tests = true
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rtc_restrict_logging = false
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}
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if (is_ios) {
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rtc_build_libjpeg = false
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rtc_enable_protobuf = false
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}
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if (current_cpu == "arm") {
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rtc_prefer_fixed_point = true
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}
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# TODO(ljubomir): Unset rtc_use_openmax_dl for mips64el once mips64el gets
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# supported in GN (since openmax_dl is not supported for mips64el).
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if (!is_ios && (current_cpu != "arm" || arm_version >= 7)) {
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rtc_use_openmax_dl = true
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} else {
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rtc_use_openmax_dl = false
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}
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# WebRTC builds ARM v7 Neon instruction set optimized code for both iOS and
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# Android, which is why we currently cannot use the variables in
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# //build/config/arm.gni (since it disables Neon for Android).
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rtc_build_armv7_neon = (current_cpu == "arm" && arm_version >= 7)
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}
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# Make it possible to provide custom locations for some libraries (move these
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# up into declare_args should we need to actually use them for the GN build).
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rtc_libvpx_dir = "//third_party/libvpx"
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rtc_libyuv_dir = "//third_party/libyuv"
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rtc_opus_dir = "//third_party/opus"
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