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webrtc/webrtc.gni
Anders Carlsson dd8c16574e Enable building WebRTC without built-in software codecs 
This CL adds a GN build flag to include builtin software codecs
(enabled by default).

When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.

Replaces https://webrtc-review.googlesource.com/c/src/+/29203

Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
2018-01-31 08:33:59 +00:00

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# 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/config/sanitizers/sanitizers.gni")
import("//build/config/ui.gni")
import("//build_overrides/build.gni")
if (!build_with_chromium && is_component_build) {
print("The Gn argument `is_component_build` is currently " +
"ignored for WebRTC builds.")
print("Component builds are supported by Chromium and the argument " +
"`is_component_build` makes it possible to create shared libraries " +
"instead of static libraries.")
print("If an app depends on WebRTC it makes sense to just depend on the " +
"WebRTC static library, so there is no difference between " +
"`is_component_build=true` and `is_component_build=false`.")
print(
"More info about component builds at: " + "https://chromium.googlesource.com/chromium/src/+/master/docs/component_build.md")
assert(!is_component_build, "Component builds are not supported in WebRTC.")
}
if (is_ios) {
import("//build/config/ios/rules.gni")
}
declare_args() {
# Include the iLBC audio codec?
rtc_include_ilbc = true
# Disable this to avoid building the Opus audio codec.
rtc_include_opus = true
# Enable this if the Opus version upon which WebRTC is built supports direct
# encoding of 120 ms packets.
rtc_opus_support_120ms_ptime = true
# Enable this to let the Opus audio codec change complexity on the fly.
rtc_opus_variable_complexity = false
# 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
# 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 false to skip building examples.
rtc_build_examples = true
# Set this to false to skip building tools.
rtc_build_tools = true
# Set this to false to skip building code that requires X11.
rtc_use_x11 = use_x11
# 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 = ""
# Links a default implementation of task queues to targets
# that depend on the target rtc_task_queue. Set to false to
# use an external implementation.
rtc_link_task_queue_impl = true
if (current_cpu == "arm" || current_cpu == "arm64") {
rtc_prefer_fixed_point = true
}
# 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 =
is_chrome_branded && proprietary_codecs && !is_android && !is_ios
# 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, replace the audio output with a sinus tone at 440Hz.
# The ADM will ask for audio data from WebRTC but instead of reading real
# audio samples from NetEQ, a sinus tone will be generated and replace the
# real audio samples.
rtc_audio_device_plays_sinus_tone = 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
# Disable this to build without support for built-in software codecs.
rtc_use_builtin_sw_codecs = true
}
if (!build_with_mozilla) {
import("//testing/test.gni")
}
# 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() {
# Enables the use of protocol buffers for debug recordings.
rtc_enable_protobuf = !build_with_mozilla
# Set this to disable building with support for SCTP data channels.
rtc_enable_sctp = !build_with_mozilla
# Disable these to not build components which can be externally provided.
rtc_build_json = !build_with_mozilla
rtc_build_libsrtp = !build_with_mozilla
rtc_build_libvpx = !build_with_mozilla
rtc_libvpx_build_vp9 = !build_with_mozilla
rtc_build_openmax_dl = !build_with_mozilla
rtc_build_opus = !build_with_mozilla
rtc_build_ssl = !build_with_mozilla
rtc_build_usrsctp = !build_with_mozilla
# Enable libevent task queues on platforms that support it.
# rtc_link_task_queue_impl must be set to true for this to
# have an effect.
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 = !build_with_mozilla
}
if (!is_ios && (current_cpu != "arm" || arm_version >= 7) &&
current_cpu != "mips64el" && !build_with_mozilla) {
rtc_use_openmax_dl = true
} else {
rtc_use_openmax_dl = false
}
# 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 && !build_with_mozilla
rtc_restrict_logging = build_with_chromium || build_with_mozilla
# 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 && !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_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 && rtc_use_x11)
###############################################################################
# Templates
#
# Points to // 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" ]
if (is_mac || is_ios) {
rtc_common_configs += [ "//build/config/compiler:enable_arc" ]
}
# 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 = []
}
webrtc_default_visibility = [ webrtc_root + "/*" ]
if (build_with_chromium) {
# Allow Chromium's WebRTC overrides targets to bypass the regular
# visibility restrictions.
webrtc_default_visibility += [ webrtc_root + "/../webrtc_overrides/*" ]
}
template("rtc_test") {
test(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Always override to public because when target_os is Android the `test`
# template can override it to [ "*" ] and we want to avoid conditional
# visibility.
visibility = [ "*" ]
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
}
if (!build_with_chromium && is_android) {
android_manifest = webrtc_root + "test/android/AndroidManifest.xml"
deps += [ webrtc_root + "test:native_test_java" ]
}
}
}
template("rtc_source_set") {
source_set(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
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",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
configs += invoker.configs
configs -= rtc_remove_configs
configs -= invoker.suppressed_configs
deps = [
"//build/config:exe_and_shlib_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",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
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",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
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
}
}
}
if (is_ios) {
set_defaults("rtc_ios_xctest_test") {
configs = rtc_add_configs
suppressed_configs = []
}
template("rtc_ios_xctest_test") {
ios_xctest_test(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
forward_variables_from(invoker, [ "visibility" ])
if (!defined(visibility)) {
visibility = webrtc_default_visibility
}
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("ios_framework_bundle_with_umbrella_header") {
forward_variables_from(invoker, [ "output_name" ])
umbrella_header_path =
"$target_gen_dir/$output_name.framework/Headers/$output_name.h"
ios_framework_bundle(target_name) {
forward_variables_from(invoker, "*", [])
deps += [ ":copy_umbrella_header_$target_name" ]
}
action("umbrella_header_$target_name") {
forward_variables_from(invoker, [ "public_headers" ])
script = "//tools_webrtc/ios/generate_umbrella_header.py"
outputs = [
umbrella_header_path,
]
args = [
"--out",
rebase_path(umbrella_header_path, root_build_dir),
"--sources",
] + rebase_path(public_headers, "objc/Framework/Headers/")
}
copy("copy_umbrella_header_$target_name") {
sources = [
umbrella_header_path,
]
outputs = [
"$root_out_dir/$output_name.framework/Headers/$output_name.h",
]
deps = [
":umbrella_header_$target_name",
]
}
}
set_defaults("ios_framework_bundle_with_umbrella_header") {
configs = default_shared_library_configs
}
}
if (is_android) {
template("rtc_android_library") {
android_library(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Treat warnings as errors.
javac_args = [ "-Werror" ]
# TODO(sakal): Fix build hooks crbug.com/webrtc/8168
no_build_hooks = true
}
}
template("rtc_android_apk") {
android_apk(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Treat warnings as errors.
javac_args = [ "-Werror" ]
# TODO(sakal): Fix build hooks crbug.com/webrtc/8168
no_build_hooks = true
}
}
template("rtc_instrumentation_test_apk") {
instrumentation_test_apk(target_name) {
forward_variables_from(invoker,
"*",
[
"configs",
"public_configs",
"suppressed_configs",
"visibility",
])
# Treat warnings as errors.
javac_args = [ "-Werror" ]
# TODO(sakal): Fix build hooks crbug.com/webrtc/8168
no_build_hooks = true
}
}
}
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