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Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )

Browse Source 
Reason for revert:
Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file)

Original issue's description:
> Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
>
> Reason for revert:
> This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio
>
> Original issue's description:
> > Moving webrtc.gni up one level from build/
> >
> > BUG=webrtc:7030
> >
> > Review-Url: https://codereview.webrtc.org/2651543003
> > Cr-Commit-Position: refs/heads/master@{#16241}
> > Committed: 35a32700fc
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2657563002
> Cr-Commit-Position: refs/heads/master@{#16244}
> Committed: 69dc7dbe24

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2654773002
Cr-Commit-Position: refs/heads/master@{#16247}
This commit is contained in:
mbonadei 2017-01-24 06:58:22 -08:00 committed by Commit bot
parent b54c63ffb0
commit 9aa3f0a200
43 changed files with 368 additions and 359 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
BUILD.gn
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@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("webrtc/build/webrtc.gni")
import("webrtc/webrtc.gni")
group("default") {
testonly = true

2
webrtc/BUILD.gn
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@ -10,7 +10,7 @@
import("//build/config/linux/pkg_config.gni")
import("//build/config/sanitizers/sanitizers.gni")
import("build/webrtc.gni")
import("webrtc.gni")
import("//third_party/protobuf/proto_library.gni")
if (is_android) {
import("//build/config/android/config.gni")

2
webrtc/api/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")

2
webrtc/audio/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
rtc_static_library("audio") {
sources = [

2
webrtc/audio/utility/BUILD.gn
View File

@ -5,7 +5,7 @@
# 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/webrtc.gni")
import("../../webrtc.gni")
group("utility") {
public_deps = [

2
webrtc/base/BUILD.gn
View File

@ -8,7 +8,7 @@
import("//build/config/crypto.gni")
import("//build/config/ui.gni")
import("../build/webrtc.gni")
import("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")

320
webrtc/build/webrtc.gni
View File

@ -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")

2
webrtc/call/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
rtc_source_set("call_interfaces") {
sources = [

2
webrtc/common_audio/BUILD.gn
View File

@ -7,7 +7,7 @@
# be found in the AUTHORS file in the root of the source tree.
import("//build/config/arm.gni")
import("../build/webrtc.gni")
import("../webrtc.gni")
config("common_audio_config") {
include_dirs = [

2
webrtc/common_video/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
config("common_video_config") {
include_dirs = [

2
webrtc/examples/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")

2
webrtc/logging/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
import("//third_party/protobuf/proto_library.gni")
if (is_android) {
import("//build/config/android/config.gni")

2
webrtc/media/BUILD.gn
View File

@ -7,7 +7,7 @@
# be found in the AUTHORS file in the root of the source tree.
import("//build/config/linux/pkg_config.gni")
import("../build/webrtc.gni")
import("../webrtc.gni")
group("media") {
public_deps = [

2
webrtc/modules/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
import("audio_coding/audio_coding.gni")
group("modules") {

2
webrtc/modules/audio_coding/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
import("audio_coding.gni")
import("//build/config/arm.gni")
import("//third_party/protobuf/proto_library.gni")

2
webrtc/modules/audio_coding/audio_coding.gni
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
audio_codec_defines = []
if (rtc_include_ilbc) {

2
webrtc/modules/audio_conference_mixer/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
config("audio_conference_mixer_config") {
visibility = [ ":*" ] # Only targets in this file can depend on this.

2
webrtc/modules/audio_device/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")

2
webrtc/modules/audio_mixer/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
group("audio_mixer") {
public_deps = [

2
webrtc/modules/audio_processing/BUILD.gn
View File

@ -8,7 +8,7 @@
import("//build/config/arm.gni")
import("//third_party/protobuf/proto_library.gni")
import("../../build/webrtc.gni")
import("../../webrtc.gni")
declare_args() {
# Disables the usual mode where we trust the reported system delay

2
webrtc/modules/bitrate_controller/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("bitrate_controller") {
# TODO(mbonadei): Remove (bugs.webrtc.org/6828)

2
webrtc/modules/congestion_controller/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("congestion_controller") {
sources = [

2
webrtc/modules/desktop_capture/BUILD.gn
View File

@ -7,7 +7,7 @@
# be found in the AUTHORS file in the root of the source tree.
import("//build/config/ui.gni")
import("../../build/webrtc.gni")
import("../../webrtc.gni")
use_desktop_capture_differ_sse2 = current_cpu == "x86" || current_cpu == "x64"

2
webrtc/modules/media_file/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
config("media_file_config") {
visibility = [ ":*" ] # Only targets in this file can depend on this.

2
webrtc/modules/pacing/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("pacing") {
sources = [

2
webrtc/modules/remote_bitrate_estimator/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("remote_bitrate_estimator") {
# TODO(mbonadei): Remove (bugs.webrtc.org/6828)

2
webrtc/modules/rtp_rtcp/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("rtp_rtcp") {
sources = [

2
webrtc/modules/utility/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("utility") {
sources = [

2
webrtc/modules/video_capture/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
# Note this target is missing an implementation for the video capture.
# Targets must link with either 'video_capture' or

2
webrtc/modules/video_coding/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
rtc_static_library("video_coding") {
sources = [

2
webrtc/modules/video_processing/BUILD.gn
View File

@ -7,7 +7,7 @@
# be found in the AUTHORS file in the root of the source tree.
import("//build/config/arm.gni")
import("../../build/webrtc.gni")
import("../../webrtc.gni")
build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"

2
webrtc/p2p/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
group("p2p") {
public_deps = [

2
webrtc/pc/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")

2
webrtc/sdk/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
if (is_ios) {
import("//build/config/ios/rules.gni")
}

2
webrtc/sdk/android/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("//webrtc/build/webrtc.gni")
import("//webrtc/webrtc.gni")
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")

2
webrtc/stats/BUILD.gn
View File

@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
group("stats") {
public_deps = [

2
webrtc/system_wrappers/BUILD.gn
View File

@ -10,7 +10,7 @@ if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
import("../build/webrtc.gni")
import("../webrtc.gni")
rtc_static_library("system_wrappers") {
sources = [

2
webrtc/test/BUILD.gn
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@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
import("//build/config/ui.gni")
if (is_android) {
import("//build/config/android/rules.gni")

2
webrtc/test/fuzzers/BUILD.gn
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@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../build/webrtc.gni")
import("../../webrtc.gni")
import("//build/config/features.gni")
import("//testing/libfuzzer/fuzzer_test.gni")

2
webrtc/tools/BUILD.gn
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@ -7,7 +7,7 @@
# be found in the AUTHORS file in the root of the source tree.
import("//third_party/protobuf/proto_library.gni")
import("../build/webrtc.gni")
import("../webrtc.gni")
group("tools") {
# This target shall build all targets in tools/.

2
webrtc/video/BUILD.gn
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@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
rtc_static_library("video") {
sources = [

2
webrtc/voice_engine/BUILD.gn
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@ -6,7 +6,7 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../build/webrtc.gni")
import("../webrtc.gni")
rtc_static_library("audio_coder") {
sources = [

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webrtc/webrtc.gni Normal file
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@ -0,0 +1,325 @@
# 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 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
}
}
}