Files
platform-external-webrtc/ortc/BUILD.gn
Mirko Bonadei 5af3051f84 Fix no_exit_time_destructors in ortc.
Non trivially destructible objects with static storage are disallowed
by the style guide.

This CL just removes 'static' since these objects are constructed once
or twice in the entire application.

Bug: webrtc:9693
Change-Id: I7509e2c088dd5ec0ac13f08053ecb76cf8259d90
Reviewed-on: https://webrtc-review.googlesource.com/98840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24634}
2018-09-07 20:47:15 +00:00

111 lines
3.4 KiB
Plaintext

# 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("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
rtc_static_library("ortc") {
defines = []
sources = [
"ortcfactory.cc",
"ortcfactory.h",
"ortcrtpreceiveradapter.cc",
"ortcrtpreceiveradapter.h",
"ortcrtpsenderadapter.cc",
"ortcrtpsenderadapter.h",
"rtptransportadapter.cc",
"rtptransportadapter.h",
"rtptransportcontrolleradapter.cc",
"rtptransportcontrolleradapter.h",
]
# TODO(deadbeef): Create a separate target for the common things ORTC and
# PeerConnection code shares, so that ortc can depend on that instead of
# libjingle_peerconnection.
deps = [
"../api:libjingle_peerconnection_api",
"../api:ortc_api",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../call:call_interfaces",
"../call:rtp_sender",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_audio_video",
"../media:rtc_media",
"../media:rtc_media_base",
"../modules/audio_processing:audio_processing",
"../p2p:rtc_p2p",
"../pc:libjingle_peerconnection",
"../pc:peerconnection",
"../pc:rtc_pc",
"../pc:rtc_pc_base",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_include_tests) {
rtc_test("ortc_unittests") {
testonly = true
sources = [
"ortcfactory_integrationtest.cc",
"ortcfactory_unittest.cc",
"ortcrtpreceiver_unittest.cc",
"ortcrtpsender_unittest.cc",
"rtptransport_unittest.cc",
"rtptransportcontroller_unittest.cc",
"srtptransport_unittest.cc",
"testrtpparameters.cc",
"testrtpparameters.h",
]
deps = [
":ortc",
"../api:libjingle_peerconnection_api",
"../api:ortc_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../media:rtc_media_tests_utils",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../pc:pc_test_utils",
"../pc:peerconnection",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base/system:arch",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
}
}
}