| # 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", |
| ] |
| } |
| } |