31a3397e96
Merge conflicts resolved in: * api/neteq/neteq.h * audio/channel_receive.cc * audio/channel_receive.h * media/engine/simulcast_encoder_adapter.cc * modules/audio_coding/BUILD.gn * modules/audio_coding/acm2/acm_resampler.cc * modules/audio_coding/acm2/acm_resampler_unittest.cc * modules/audio_processing/audio_processing_impl.h * modules/video_coding/BUILD.gn * p2p/base/p2p_transport_channel_unittest.cc * pc/jsep_transport_controller.cc * pc/rtp_transport.h * pc/rtp_transport_internal.h * unsafe_buffers_paths.txt
105 lines
3.7 KiB
Plaintext
105 lines
3.7 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/dotfile_settings.gni")
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# The location of the build configuration file.
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buildconfig = "//build/config/BUILDCONFIG.gn"
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# The python interpreter to use by default. On Windows, this will look
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# for python3.exe and python3.bat.
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script_executable = "python3"
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# The secondary source root is a parallel directory tree where
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# GN build files are placed when they can not be placed directly
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# in the source tree, e.g. for third party source trees.
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secondary_source = "//build/secondary/"
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# These are the targets to skip header checking by default. The files in targets
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# matching these patterns (see "gn help label_pattern" for format) will not have
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# their includes checked for proper dependencies when you run either
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# "gn check" or "gn gen --check".
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no_check_targets = [
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"//third_party/icu/*",
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# TODO: crbug/326607005 - GTEST_HAS_ABSL is broken
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"//third_party/googletest:gmock",
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"//third_party/googletest:gtest",
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]
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# These are the list of GN files that run exec_script. This allowlist exists
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# to force additional review for new uses of exec_script, which is strongly
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# discouraged except for gypi_to_gn calls.
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exec_script_allowlist = build_dotfile_settings.exec_script_allowlist +
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[ "//build_overrides/build.gni" ]
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export_compile_commands = [ "*" ]
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default_args = {
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# Webrtc does not support component builds because we are not using the
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# template "component" but we rely directly on "rtc_static_library" and
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# "rtc_shared_library". This means that we cannot use the chromium default
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# value for this argument.
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# This also means that the user can override this value using --args or
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# the args.gn file but this setting will be ignored because we don't support
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# component builds.
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is_component_build = false
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# RingRTC change to control MacOS/iOS targets
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mac_sdk_min = "10.12.1"
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ios_deployment_target = "14.0"
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# The SDK API level, in contrast, is set by build/android/AndroidManifest.xml.
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android_ndk_api_level = 23
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# WebRTC does not provide the gflags dependency. Because libyuv uses it only
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# for unittests, it can be disabled (see third_party/libyuv/BUILD.gn)
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libyuv_use_gflags = false
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# RingRTC change to disable AV1
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enable_libaom = false
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gtest_enable_absl_printers = true
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# Differently from Chromium, WebRTC still support SDK 21.
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default_min_sdk_version = 23
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# Prevent jsoncpp to pass -Wno-deprecated-declarations to users
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jsoncpp_no_deprecated_declarations = false
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# Fixes the abi-revision issue.
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# TODO(https://bugs.webrtc.org/14437): Remove this section if general
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# Chromium fix resolves the problem.
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fuchsia_sdk_readelf_exec =
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"//third_party/llvm-build/Release+Asserts/bin/llvm-readelf"
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# WebRTC doesn't use jni_zero's multiplexing. Since this causes an error
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# let's temporarily disable it.
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enable_jni_multiplexing = false
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# use_fuzztest_wrapper adds a dependency to //base so we have to disable it.
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use_fuzztest_wrapper = false
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# Enable Rust in WebRTC
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enable_rust = true
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enable_rust_cxx = true
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enable_chromium_prelude = true
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# See context on https://issues.chromium.org/issues/415076048.
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enable_js_protobuf = false
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# Use Siso instead of Ninja.
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use_siso = true
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clang_unsafe_buffers_paths = "//unsafe_buffers_paths.txt"
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# WebRTC must stay in C++20 for longer compared to Chromium.
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use_cxx23 = false
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}
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