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WebRTC编译系统之gn files
在“WebRTC 构建系统介绍之gn和ninja”中，大概介绍了 gn 和 ninja 的简单用法，这次来看看 gn 用到的项目文件 .gn 、 .gni 和 DEPS ，它们指导了如何生成 ninja 构建文件。
借用 C++ 的概念，如果把 gn 看成一个编译系统， .gn 就是源文件， .gni 就是头文件。我们姑且这么理解就好了（其实 gni 里做的事情， gn 都可以做）。DEPS 主要用来设定包含路径。
gn 和 gni 文件都在源码树中，比如 src 目录。当执行 gn gen 时，gn 工具根据 gn 和 gni 生成 ninja 文件并将这些 ninja 文件放到指定的构建目录中。
.gn
.gn 文件是 GN build 的 “源文件”，在这里可以做各种条件判断和配置，gn 会根据这些配置生成特定的 ninja 文件。
.gn 文件中可以使用预定义的参数，比如 is_debug ， target_os ， rtc_use_h264 等。
.gn 中可以 import .gni 文件。
看一下 src/BUILD.gn ：
import("webrtc/webrtc.gni")
group("default") {
testonly = true
deps = [
"//webrtc",
"//webrtc/examples",
"//webrtc/tools",
]
if (rtc_include_tests) {
deps += [ "//webrtc:webrtc_tests" ]
}
}
.gn 和 .gni 文件中用到各种指令，都在这里有说明：GN Reference。
这个 gn 文件中，导入了 webrtc/webrtc.gni 文件。
这个 gn 文件，用 group 指令声明了一个 default 目标，这个目标依赖 webrtc 、 webrtc/examples 和 webrtc/tools ，你可以在 webrtc 、 webrtc/examples 、 webrtc/tools 目录下找到对应的 BUILD.gn 。你可以把 group 当做 VS 的 solution ，或者 QtCreator 的 dir 项目。
gn 文件中也可以通过 defines 来定义宏，通过 cflags 来指定传递给编译器的标记，通过 ldflags 指定传递给链接器的标记，还可以使用 sources 指定源文件。下面是 webrtc/BUILD.gn 文件的部分内容：
if (is_win) {
defines += [
"WEBRTC_WIN",
"_CRT_SECURE_NO_WARNINGS", # Suppress warnings about _vsnprinf
]
}
if (is_android) {
defines += [
"WEBRTC_LINUX",
"WEBRTC_ANDROID",
]
}
if (is_chromeos) {
defines += [ "CHROMEOS" ]
}
if (rtc_sanitize_coverage != "") {
assert(is_clang, "sanitizer coverage requires clang")
cflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
ldflags += [ "-fsanitize-coverage=${rtc_sanitize_coverage}" ]
}
.gni
gni 文件是 GN build 使用的头文件，它里面可以做各种事情，比如定义变量、宏、定义配置、定义模板等。
看下 webrtc/webrtc.gni 文件：
# 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_overrides/build.gni")
import("//testing/test.gni")
declare_args() {
# 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 = false
# 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
}
}
}
webrtc.gni 是一个比较特殊的 gni 文件，你可以把它看做全局配置文件。
webrtc.gni 定义了 WebRTC 项目用到的一些标记，比如 rtc_build_libvpx、rtc_build_ssl、rtc_use_h264 等。
还使用 template 语句定义了几个模板，比如 rtc_executable 、 rtc_static_library 、 rtc_shared_library ，这几个模板定义了生成可执行文件、静态库、动态库的规则。在 webrtc/examples/BUILD.gn 中就有用到这些模板，用它们来指导如何生成可执行文件、静态库等。
你也可以直接使用 gn 内置的 shared_library 和 static_library 来声明目标，比如 third_party/ffmpeg/BUILD.gn 就使用 shared_library 来生成动态库。
DEPS 文件
给个例子看看吧，webrtc/examples/DEPS ：
include_rules = [
"+WebRTC",
"+webrtc/api",
"+webrtc/base",
"+webrtc/media",
"+webrtc/modules/audio_device",
"+webrtc/modules/video_capture",
"+webrtc/p2p",
"+webrtc/pc",
]
include_rules 定义了包含路径。
修改 .gn 和 .gni
了解 .gn 和 .gni 文件的目的是修改它们。比如你想打开 WebRTC 对 H264 的支持，就可以修改 webrtc/webrtc.gni ，直接把 rtc_use_h264 设置为 true 。
比如你想为某个模块加一些文件，就可以修改 .gn 文件，修改 sources 变量，直接把你的源文件加进去。
好啦，到这儿吧。
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- 打开 WebRTC 的日志（native api）
- 让WebRTC支持H264编解码
- WebRTC编译系统之gn和ninja
WebRTC编译系统之gn files
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