# 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") import("//testing/test.gni") group("media") { 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", ] } } 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/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 += [ "..:common_config", ":rtc_media_warnings_config", ] public_configs = [ "..:common_inherited_config" ] if (is_clang) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). 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_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" ] } 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", ] } if (is_mac && current_cpu == "x86") { sources += [ "devices/carbonvideorenderer.cc", "devices/carbonvideorenderer.h", ] libs += [ "Carbon.framework" ] } if (is_ios || (is_mac && current_cpu != "x86")) { defines += [ "CARBON_DEPRECATED=YES" ] } deps += [ "..:webrtc_common", "../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" ] } } source_set("rtc_unittest_main") { testonly = true deps = [] sources = [ "base/fakemediaengine.h", "base/fakenetworkinterface.h", "base/fakertp.h", "base/fakevideocapturer.h", "base/fakevideorenderer.h", "base/testutils.cc", "base/testutils.h", "engine/fakewebrtccall.cc", "engine/fakewebrtccall.h", "engine/fakewebrtccommon.h", "engine/fakewebrtcdeviceinfo.h", "engine/fakewebrtcvcmfactory.h", "engine/fakewebrtcvideocapturemodule.h", "engine/fakewebrtcvideoengine.h", "engine/fakewebrtcvoiceengine.h", ] configs += [ "..:common_config", ":rtc_unittest_main_config", ] public_configs = [ "..:common_inherited_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. 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 += [ "-Wno-unused-function" ] } if (!is_win) { cflags = [ "-Wno-sign-compare" ] cflags_cc = [ "-Wno-overloaded-virtual" ] } } test("rtc_media_unittests") { testonly = true 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/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 += [ "..:common_config", ":rtc_media_unittests_config", ] public_configs = [ "..:common_inherited_config" ] 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). configs -= [ "//build/config/clang:extra_warnings", "//build/config/clang:find_bad_constructs", ] } if (is_android) { deps += [ "//testing/android/native_test:native_test_support" ] # This needs to be kept in sync with the rtc_media_unittests.isolate file. # TODO(kjellander); Move this to android_assets targets instead. data = [ "//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", ] } 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", ] } }