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Reland of Moving webrtc.gni up one level from build/ (patchset Jumpin…
…gYang001#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 JumpingYang001#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 > > [email protected] > # 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 [email protected] # 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}
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -1,325 +1,9 @@ | ||
| # 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 | ||
| # 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 | ||
| } | ||
| } | ||
| } | ||
| import("../webrtc.gni") |
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