From 9aa3f0a2006dfc7278e0c93cd073d4bbb190b796 Mon Sep 17 00:00:00 2001 From: mbonadei Date: Tue, 24 Jan 2017 06:58:22 -0800 Subject: [PATCH] Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) 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: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > 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: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 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} --- BUILD.gn | 2 +- webrtc/BUILD.gn | 2 +- webrtc/api/BUILD.gn | 2 +- webrtc/audio/BUILD.gn | 2 +- webrtc/audio/utility/BUILD.gn | 2 +- webrtc/base/BUILD.gn | 2 +- webrtc/build/webrtc.gni | 320 +---------------- webrtc/call/BUILD.gn | 2 +- webrtc/common_audio/BUILD.gn | 2 +- webrtc/common_video/BUILD.gn | 2 +- webrtc/examples/BUILD.gn | 2 +- webrtc/logging/BUILD.gn | 2 +- webrtc/media/BUILD.gn | 2 +- webrtc/modules/BUILD.gn | 2 +- webrtc/modules/audio_coding/BUILD.gn | 2 +- webrtc/modules/audio_coding/audio_coding.gni | 2 +- .../modules/audio_conference_mixer/BUILD.gn | 2 +- webrtc/modules/audio_device/BUILD.gn | 2 +- webrtc/modules/audio_mixer/BUILD.gn | 2 +- webrtc/modules/audio_processing/BUILD.gn | 2 +- webrtc/modules/bitrate_controller/BUILD.gn | 2 +- webrtc/modules/congestion_controller/BUILD.gn | 2 +- webrtc/modules/desktop_capture/BUILD.gn | 2 +- webrtc/modules/media_file/BUILD.gn | 2 +- webrtc/modules/pacing/BUILD.gn | 2 +- .../modules/remote_bitrate_estimator/BUILD.gn | 2 +- webrtc/modules/rtp_rtcp/BUILD.gn | 2 +- webrtc/modules/utility/BUILD.gn | 2 +- webrtc/modules/video_capture/BUILD.gn | 2 +- webrtc/modules/video_coding/BUILD.gn | 2 +- webrtc/modules/video_processing/BUILD.gn | 2 +- webrtc/p2p/BUILD.gn | 2 +- webrtc/pc/BUILD.gn | 2 +- webrtc/sdk/BUILD.gn | 2 +- webrtc/sdk/android/BUILD.gn | 2 +- webrtc/stats/BUILD.gn | 2 +- webrtc/system_wrappers/BUILD.gn | 2 +- webrtc/test/BUILD.gn | 2 +- webrtc/test/fuzzers/BUILD.gn | 2 +- webrtc/tools/BUILD.gn | 2 +- webrtc/video/BUILD.gn | 2 +- webrtc/voice_engine/BUILD.gn | 2 +- webrtc/webrtc.gni | 325 ++++++++++++++++++ 43 files changed, 368 insertions(+), 359 deletions(-) create mode 100644 webrtc/webrtc.gni diff --git a/BUILD.gn b/BUILD.gn index 62e06ad4ac..4e6816fd85 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -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 diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index c3ff51de53..d99dc4da4a 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -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") diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn index af49219e46..038766aa9a 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/api/BUILD.gn @@ -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") diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn index 9ded96f06f..80d541624c 100644 --- a/webrtc/audio/BUILD.gn +++ b/webrtc/audio/BUILD.gn @@ -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 = [ diff --git a/webrtc/audio/utility/BUILD.gn b/webrtc/audio/utility/BUILD.gn index df5f322fed..2ef5eba338 100644 --- a/webrtc/audio/utility/BUILD.gn +++ b/webrtc/audio/utility/BUILD.gn @@ -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 = [ diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn index 248ef13f5f..13ad5fdc00 100644 --- a/webrtc/base/BUILD.gn +++ b/webrtc/base/BUILD.gn @@ -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") diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni index d179ed4e7d..04bab4677e 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") diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn index 195c37f614..acad72d461 100644 --- a/webrtc/call/BUILD.gn +++ b/webrtc/call/BUILD.gn @@ -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 = [ diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn index 2d46f67ef2..a3cf046bb4 100644 --- a/webrtc/common_audio/BUILD.gn +++ b/webrtc/common_audio/BUILD.gn @@ -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 = [ diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn index 3b33bcca8d..1c13fa4b06 100644 --- a/webrtc/common_video/BUILD.gn +++ b/webrtc/common_video/BUILD.gn @@ -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 = [ diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn index 6f7eaa702b..dfe15d18d7 100644 --- a/webrtc/examples/BUILD.gn +++ b/webrtc/examples/BUILD.gn @@ -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") diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn index 792eb93a9a..fd0caca037 100644 --- a/webrtc/logging/BUILD.gn +++ b/webrtc/logging/BUILD.gn @@ -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") diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn index 7f8a1f6fca..06aed45662 100644 --- a/webrtc/media/BUILD.gn +++ b/webrtc/media/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index d87e7f9612..e44b35f4c3 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -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") { diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 044d57e6a0..404637357d 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -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") diff --git a/webrtc/modules/audio_coding/audio_coding.gni b/webrtc/modules/audio_coding/audio_coding.gni index 13577f86b3..0f3a75fbac 100644 --- a/webrtc/modules/audio_coding/audio_coding.gni +++ b/webrtc/modules/audio_coding/audio_coding.gni @@ -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) { diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn index b4705d5329..428bcac03c 100644 --- a/webrtc/modules/audio_conference_mixer/BUILD.gn +++ b/webrtc/modules/audio_conference_mixer/BUILD.gn @@ -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. diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn index 4dc6760144..ac8fdbe959 100644 --- a/webrtc/modules/audio_device/BUILD.gn +++ b/webrtc/modules/audio_device/BUILD.gn @@ -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") diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn index 25c782bc9b..083e02b9ca 100644 --- a/webrtc/modules/audio_mixer/BUILD.gn +++ b/webrtc/modules/audio_mixer/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn index 4f3ea13efb..87501f4cdb 100644 --- a/webrtc/modules/audio_processing/BUILD.gn +++ b/webrtc/modules/audio_processing/BUILD.gn @@ -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 diff --git a/webrtc/modules/bitrate_controller/BUILD.gn b/webrtc/modules/bitrate_controller/BUILD.gn index b2747db26c..58398f367f 100644 --- a/webrtc/modules/bitrate_controller/BUILD.gn +++ b/webrtc/modules/bitrate_controller/BUILD.gn @@ -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) diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn index 03e0d5c93f..a20339cd78 100644 --- a/webrtc/modules/congestion_controller/BUILD.gn +++ b/webrtc/modules/congestion_controller/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/desktop_capture/BUILD.gn b/webrtc/modules/desktop_capture/BUILD.gn index 67ec8cea37..9c9a46631e 100644 --- a/webrtc/modules/desktop_capture/BUILD.gn +++ b/webrtc/modules/desktop_capture/BUILD.gn @@ -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" diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn index 8003cd8160..32825afb2b 100644 --- a/webrtc/modules/media_file/BUILD.gn +++ b/webrtc/modules/media_file/BUILD.gn @@ -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. diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn index 0a357c00d0..f94def78be 100644 --- a/webrtc/modules/pacing/BUILD.gn +++ b/webrtc/modules/pacing/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn index 81028f6b63..13e2382b84 100644 --- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn +++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn @@ -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) diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 9b236218b3..4fa4d4963d 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index 5b6a232adc..ec11b9fdf1 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/video_capture/BUILD.gn b/webrtc/modules/video_capture/BUILD.gn index 6887552b58..90b824a5b8 100644 --- a/webrtc/modules/video_capture/BUILD.gn +++ b/webrtc/modules/video_capture/BUILD.gn @@ -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 diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn index f84936ad79..5750d10178 100644 --- a/webrtc/modules/video_coding/BUILD.gn +++ b/webrtc/modules/video_coding/BUILD.gn @@ -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 = [ diff --git a/webrtc/modules/video_processing/BUILD.gn b/webrtc/modules/video_processing/BUILD.gn index 5da0ae89ee..6da43b422b 100644 --- a/webrtc/modules/video_processing/BUILD.gn +++ b/webrtc/modules/video_processing/BUILD.gn @@ -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" diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn index e99440cf43..04e9b17908 100644 --- a/webrtc/p2p/BUILD.gn +++ b/webrtc/p2p/BUILD.gn @@ -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 = [ diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn index d447db743a..5aa3ae4eac 100644 --- a/webrtc/pc/BUILD.gn +++ b/webrtc/pc/BUILD.gn @@ -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") diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn index e610242f63..9f3ccc6b7d 100644 --- a/webrtc/sdk/BUILD.gn +++ b/webrtc/sdk/BUILD.gn @@ -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") } diff --git a/webrtc/sdk/android/BUILD.gn b/webrtc/sdk/android/BUILD.gn index 1df43e2e2a..710d962125 100644 --- a/webrtc/sdk/android/BUILD.gn +++ b/webrtc/sdk/android/BUILD.gn @@ -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") diff --git a/webrtc/stats/BUILD.gn b/webrtc/stats/BUILD.gn index 8fa80879a3..85ba21d15e 100644 --- a/webrtc/stats/BUILD.gn +++ b/webrtc/stats/BUILD.gn @@ -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 = [ diff --git a/webrtc/system_wrappers/BUILD.gn b/webrtc/system_wrappers/BUILD.gn index 745867bf91..96177fd508 100644 --- a/webrtc/system_wrappers/BUILD.gn +++ b/webrtc/system_wrappers/BUILD.gn @@ -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 = [ diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn index 5ad85fce24..9893814748 100644 --- a/webrtc/test/BUILD.gn +++ b/webrtc/test/BUILD.gn @@ -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") diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn index ea2352a82c..031ed8b342 100644 --- a/webrtc/test/fuzzers/BUILD.gn +++ b/webrtc/test/fuzzers/BUILD.gn @@ -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") diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn index 8773cfc377..46a0ede844 100644 --- a/webrtc/tools/BUILD.gn +++ b/webrtc/tools/BUILD.gn @@ -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/. diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index ed69526d52..48dff5ba07 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -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 = [ diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index c29e646e18..cb97693232 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -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 = [ diff --git a/webrtc/webrtc.gni b/webrtc/webrtc.gni new file mode 100644 index 0000000000..53dcae2b7c --- /dev/null +++ b/webrtc/webrtc.gni @@ -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 + } + } +}