blob: 6179f67d6bfa3cd4e5296b847a334e7197005470 [file] [log] [blame]
# Copyright (c) 2016 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/linux/pkg_config.gni")
import("../build/webrtc.gni")
group("media") {
public_deps = [
":rtc_media",
]
}
config("rtc_media_defines_config") {
defines = [
"HAVE_WEBRTC_VIDEO",
"HAVE_WEBRTC_VOICE",
]
}
config("rtc_media_warnings_config") {
# GN orders flags on a target before flags from configs. The default config
# adds these flags so to cancel them out they need to come from a config and
# cannot be on the target directly.
if (!is_win) {
cflags = [ "-Wno-deprecated-declarations" ]
cflags_cc = [ "-Wno-overloaded-virtual" ]
}
}
if (is_linux && rtc_use_gtk) {
pkg_config("gtk-lib") {
packages = [
"gobject-2.0",
"gthread-2.0",
"gtk+-2.0",
]
}
}
rtc_source_set("rtc_media") {
defines = []
libs = []
deps = []
sources = [
"base/audiosource.h",
"base/codec.cc",
"base/codec.h",
"base/cpuid.cc",
"base/cpuid.h",
"base/cryptoparams.h",
"base/device.h",
"base/fakescreencapturerfactory.h",
"base/hybriddataengine.h",
"base/mediachannel.h",
"base/mediacommon.h",
"base/mediaconstants.cc",
"base/mediaconstants.h",
"base/mediaengine.cc",
"base/mediaengine.h",
"base/rtpdataengine.cc",
"base/rtpdataengine.h",
"base/rtpdump.cc",
"base/rtpdump.h",
"base/rtputils.cc",
"base/rtputils.h",
"base/screencastid.h",
"base/streamparams.cc",
"base/streamparams.h",
"base/turnutils.cc",
"base/turnutils.h",
"base/videoadapter.cc",
"base/videoadapter.h",
"base/videobroadcaster.cc",
"base/videobroadcaster.h",
"base/videocapturer.cc",
"base/videocapturer.h",
"base/videocapturerfactory.h",
"base/videocommon.cc",
"base/videocommon.h",
"base/videoframe.cc",
"base/videoframe.h",
"base/videoframefactory.cc",
"base/videoframefactory.h",
"base/videorenderer.h",
"base/videosourcebase.cc",
"base/videosourcebase.h",
"devices/videorendererfactory.h",
"engine/nullwebrtcvideoengine.h",
"engine/payload_type_mapper.cc",
"engine/payload_type_mapper.h",
"engine/simulcast.cc",
"engine/simulcast.h",
"engine/webrtccommon.h",
"engine/webrtcmediaengine.cc",
"engine/webrtcmediaengine.h",
"engine/webrtcvideocapturer.cc",
"engine/webrtcvideocapturer.h",
"engine/webrtcvideocapturerfactory.cc",
"engine/webrtcvideocapturerfactory.h",
"engine/webrtcvideodecoderfactory.h",
"engine/webrtcvideoencoderfactory.h",
"engine/webrtcvideoengine2.cc",
"engine/webrtcvideoengine2.h",
"engine/webrtcvideoframe.cc",
"engine/webrtcvideoframe.h",
"engine/webrtcvideoframefactory.cc",
"engine/webrtcvideoframefactory.h",
"engine/webrtcvoe.h",
"engine/webrtcvoiceengine.cc",
"engine/webrtcvoiceengine.h",
"sctp/sctpdataengine.cc",
"sctp/sctpdataengine.h",
]
configs += [ ":rtc_media_warnings_config" ]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [
"//build/config/clang:extra_warnings",
"//build/config/clang:find_bad_constructs",
]
}
if (is_win) {
cflags = [
"/wd4245", # conversion from "int" to "size_t", signed/unsigned mismatch.
"/wd4267", # conversion from "size_t" to "int", possible loss of data.
"/wd4389", # signed/unsigned mismatch.
]
}
if (rtc_enable_intelligibility_enhancer) {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
} else {
defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
}
include_dirs = []
if (rtc_build_libyuv) {
deps += [ "$rtc_libyuv_dir" ]
public_deps = [
"$rtc_libyuv_dir",
]
} else {
# Need to add a directory normally exported by libyuv.
include_dirs += [ "$rtc_libyuv_dir/include" ]
}
if (rtc_build_usrsctp) {
include_dirs += [
# TODO(jiayl): move this into the public_configs of
# //third_party/usrsctp/BUILD.gn.
"//third_party/usrsctp/usrsctplib",
]
deps += [ "//third_party/usrsctp" ]
}
public_configs = []
if (build_with_chromium) {
deps += [ "../modules/video_capture:video_capture" ]
} else {
public_configs += [ ":rtc_media_defines_config" ]
deps += [ "../modules/video_capture:video_capture_internal_impl" ]
}
if (is_linux && rtc_use_gtk) {
sources += [
"devices/gtkvideorenderer.cc",
"devices/gtkvideorenderer.h",
]
public_configs += [ ":gtk-lib" ]
}
if (is_win) {
sources += [
"devices/gdivideorenderer.cc",
"devices/gdivideorenderer.h",
]
libs += [
"d3d9.lib",
"gdi32.lib",
"strmiids.lib",
]
}
deps += [
"..:webrtc_common",
"../api:call_api",
"../base:rtc_base_approved",
"../call",
"../libjingle/xmllite",
"../libjingle/xmpp",
"../modules/video_coding",
"../p2p",
"../system_wrappers",
"../video",
"../voice_engine",
]
}
if (rtc_include_tests) {
config("rtc_unittest_main_config") {
# GN orders flags on a target before flags from configs. The default config
# adds -Wall, and this flag have to be after -Wall -- so they need to
# come from a config and can"t be on the target directly.
if (is_clang && is_ios) {
cflags = [ "-Wno-unused-variable" ]
}
}
rtc_source_set("rtc_unittest_main") {
testonly = true
include_dirs = []
public_deps = []
deps = []
sources = [
"base/fakemediaengine.h",
"base/fakenetworkinterface.h",
"base/fakertp.h",
"base/fakevideocapturer.h",
"base/fakevideorenderer.h",
"base/test/mock_mediachannel.h",
"base/testutils.cc",
"base/testutils.h",
"engine/fakewebrtccall.cc",
"engine/fakewebrtccall.h",
"engine/fakewebrtcdeviceinfo.h",
"engine/fakewebrtcvcmfactory.h",
"engine/fakewebrtcvideocapturemodule.h",
"engine/fakewebrtcvideoengine.h",
"engine/fakewebrtcvoiceengine.h",
]
configs += [ ":rtc_unittest_main_config" ]
if (rtc_build_libyuv) {
deps += [ "$rtc_libyuv_dir" ]
public_deps += [ "$rtc_libyuv_dir" ]
} else {
# Need to add a directory normally exported by libyuv.
include_dirs += [ "$rtc_libyuv_dir/include" ]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps += [
"../base:rtc_base_tests_utils",
"//testing/gtest",
]
public_deps += [ "//testing/gmock" ]
}
config("rtc_media_unittests_config") {
# GN orders flags on a target before flags from configs. The default config
# adds -Wall, and this flag have to be after -Wall -- so they need to
# come from a config and can"t be on the target directly.
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.webrtc.org/3307.
if (is_clang && is_win) {
cflags = [
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=6266
# for -Wno-sign-compare
"-Wno-sign-compare",
"-Wno-unused-function",
]
}
if (!is_win) {
cflags = [ "-Wno-sign-compare" ]
cflags_cc = [ "-Wno-overloaded-virtual" ]
}
}
rtc_media_unittests_resources = [
"//resources/media/captured-320x240-2s-48.frames",
"//resources/media/faces.1280x720_P420.yuv",
"//resources/media/faces_I420.jpg",
"//resources/media/faces_I422.jpg",
"//resources/media/faces_I444.jpg",
"//resources/media/faces_I411.jpg",
"//resources/media/faces_I400.jpg",
]
if (is_ios) {
bundle_data("rtc_media_unittests_bundle_data") {
testonly = true
sources = rtc_media_unittests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
rtc_test("rtc_media_unittests") {
testonly = true
defines = []
deps = []
sources = [
"base/codec_unittest.cc",
"base/rtpdataengine_unittest.cc",
"base/rtpdump_unittest.cc",
"base/rtputils_unittest.cc",
"base/streamparams_unittest.cc",
"base/turnutils_unittest.cc",
"base/videoadapter_unittest.cc",
"base/videobroadcaster_unittest.cc",
"base/videocapturer_unittest.cc",
"base/videocommon_unittest.cc",
"base/videoengine_unittest.h",
"base/videoframe_unittest.h",
"engine/nullwebrtcvideoengine_unittest.cc",
"engine/payload_type_mapper_unittest.cc",
"engine/simulcast_unittest.cc",
"engine/webrtcmediaengine_unittest.cc",
"engine/webrtcvideocapturer_unittest.cc",
"engine/webrtcvideoengine2_unittest.cc",
"engine/webrtcvideoframe_unittest.cc",
"engine/webrtcvideoframefactory_unittest.cc",
"engine/webrtcvoiceengine_unittest.cc",
"sctp/sctpdataengine_unittest.cc",
]
configs += [ ":rtc_media_unittests_config" ]
if (rtc_use_h264) {
defines += [ "WEBRTC_USE_H264" ]
}
if (is_win) {
cflags = [
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
"/wd4373", # virtual function override.
"/wd4389", # signed/unsigned mismatch.
]
}
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [
"//build/config/clang:extra_warnings",
"//build/config/clang:find_bad_constructs",
]
}
data = rtc_media_unittests_resources
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
if (is_ios) {
deps += [ ":rtc_media_unittests_bundle_data" ]
}
deps += [
# TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
":rtc_media",
":rtc_unittest_main",
"../audio",
"../base:rtc_base_tests_utils",
"../system_wrappers:metrics_default",
]
}
}