| # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| # |
| # Use of this source code is governed by a BSD-style license |
| # that can be found in the LICENSE file in the root of the source |
| # tree. An additional intellectual property rights grant can be found |
| # in the file PATENTS. All contributing project authors may |
| # be found in the AUTHORS file in the root of the source tree. |
| |
| import("//build/config/arm.gni") |
| import("//build/config/features.gni") |
| import("//build/config/mips.gni") |
| import("//build/config/sanitizers/sanitizers.gni") |
| import("//build/config/ui.gni") |
| import("//build_overrides/build.gni") |
| import("//testing/test.gni") |
| |
| if (!build_with_chromium && is_component_build) { |
| print("The Gn argument `is_component_build` is currently " + |
| "ignored for WebRTC builds.") |
| print("Component builds are supported by Chromium and the argument " + |
| "`is_component_build` makes it possible to create shared libraries " + |
| "instead of static libraries.") |
| print("If an app depends on WebRTC it makes sense to just depend on the " + |
| "WebRTC static library, so there is no difference between " + |
| "`is_component_build=true` and `is_component_build=false`.") |
| print( |
| "More info about component builds at: " + "https://chromium.googlesource.com/chromium/src/+/master/docs/component_build.md") |
| assert(!is_component_build, "Component builds are not supported in WebRTC.") |
| } |
| |
| if (is_ios) { |
| import("//build/config/ios/rules.gni") |
| } |
| |
| declare_args() { |
| # Disable this to avoid building the Opus audio codec. |
| rtc_include_opus = true |
| |
| # Enable this if the Opus version upon which WebRTC is built supports direct |
| # encoding of 120 ms packets. |
| rtc_opus_support_120ms_ptime = true |
| |
| # Enable this to let the Opus audio codec change complexity on the fly. |
| rtc_opus_variable_complexity = false |
| |
| # Used to specify an external Jsoncpp include path when not compiling the |
| # library that comes with WebRTC (i.e. rtc_build_json == 0). |
| rtc_jsoncpp_root = "//third_party/jsoncpp/source/include" |
| |
| # Used to specify an external OpenSSL include path when not compiling the |
| # library that comes with WebRTC (i.e. rtc_build_ssl == 0). |
| rtc_ssl_root = "" |
| |
| # Selects fixed-point code where possible. |
| rtc_prefer_fixed_point = false |
| |
| # 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_json = 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 = "" |
| |
| # Links a default implementation of task queues to targets |
| # that depend on the target rtc_task_queue. Set to false to |
| # use an external implementation. |
| rtc_link_task_queue_impl = true |
| |
| # Enable libevent task queues on platforms that support it. |
| # rtc_link_task_queue_impl must be set to true for this to |
| # have an effect. |
| if (is_win || is_mac || is_ios || is_nacl) { |
| rtc_enable_libevent = false |
| rtc_build_libevent = false |
| } else { |
| rtc_enable_libevent = true |
| rtc_build_libevent = 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, replace the audio output with a sinus tone at 440Hz. |
| # The ADM will ask for audio data from WebRTC but instead of reading real |
| # audio samples from NetEQ, a sinus tone will be generated and replace the |
| # real audio samples. |
| rtc_audio_device_plays_sinus_tone = false |
| |
| # When set to true, test targets will declare the files needed to run memcheck |
| # as data dependencies. This is to enable memcheck execution on swarming bots. |
| rtc_use_memcheck = false |
| |
| # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done |
| # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must |
| # only be initialized once. Projects that initialize FFmpeg externally, such |
| # as Chromium, must turn this flag off so that WebRTC does not also |
| # initialize. |
| rtc_initialize_ffmpeg = !build_with_chromium |
| |
| # 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 && use_x11) |
| |
| ############################################################################### |
| # 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" ] |
| |
| if (is_mac || is_ios) { |
| rtc_common_configs += [ "//build/config/compiler:enable_arc" ] |
| } |
| |
| # Global public configuration that should be applied to all WebRTC targets. You |
| # normally shouldn't need to include this in your target as it's automatically |
| # included when using the rtc_* templates. It set the defines, include paths and |
| # compilation warnings that should be propagated to dependents of the targets |
| # depending on the target having this config. |
| rtc_common_inherited_config = webrtc_root + ":common_inherited_config" |
| |
| # Common configs to remove or add in all rtc targets. |
| rtc_remove_configs = [] |
| rtc_add_configs = rtc_common_configs |
| |
| set_defaults("rtc_test") { |
| configs = rtc_add_configs |
| suppressed_configs = [] |
| } |
| |
| set_defaults("rtc_source_set") { |
| configs = rtc_add_configs |
| suppressed_configs = [] |
| } |
| |
| set_defaults("rtc_executable") { |
| configs = rtc_add_configs |
| suppressed_configs = [] |
| } |
| |
| set_defaults("rtc_static_library") { |
| configs = rtc_add_configs |
| suppressed_configs = [] |
| } |
| |
| set_defaults("rtc_shared_library") { |
| configs = rtc_add_configs |
| suppressed_configs = [] |
| } |
| |
| template("rtc_test") { |
| test(target_name) { |
| forward_variables_from(invoker, |
| "*", |
| [ |
| "configs", |
| "data", |
| "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 |
| } |
| data = [] |
| if (defined(invoker.data)) { |
| data += invoker.data |
| } |
| if (!build_with_chromium && is_android) { |
| android_manifest = webrtc_root + "test/android/AndroidManifest.xml" |
| deps += [ webrtc_root + "test:native_test_java" ] |
| |
| # Needed for build/android/pylib/results/presentation/test_results_presentation.py |
| data += [ "//build/android/pylib/results/" ] |
| } |
| } |
| } |
| |
| 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:exe_and_shlib_deps", |
| ] |
| deps += invoker.deps |
| |
| public_configs = [ rtc_common_inherited_config ] |
| if (defined(invoker.public_configs)) { |
| public_configs += invoker.public_configs |
| } |
| } |
| } |
| |
| template("rtc_static_library") { |
| static_library(target_name) { |
| forward_variables_from(invoker, |
| "*", |
| [ |
| "configs", |
| "public_configs", |
| "suppressed_configs", |
| ]) |
| 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 |
| } |
| } |
| } |
| |
| if (is_ios) { |
| set_defaults("rtc_ios_xctest_test") { |
| configs = rtc_add_configs |
| suppressed_configs = [] |
| } |
| |
| template("rtc_ios_xctest_test") { |
| ios_xctest_test(target_name) { |
| forward_variables_from(invoker, |
| "*", |
| [ |
| "configs", |
| "public_configs", |
| "suppressed_configs", |
| ]) |
| 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 |
| } |
| } |
| } |
| } |