blob: 25aa1e07a7a29721873f34824630c6b31e0ca757 [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("../webrtc.gni")
group("media") {
public_deps = [
":rtc_media",
":rtc_media_base",
]
}
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" ]
}
}
rtc_source_set("rtc_h264_profile_id") {
sources = [
"base/h264_profile_level_id.cc",
"base/h264_profile_level_id.h",
]
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" ]
}
deps = [
"..:webrtc_common",
"../api:optional",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
]
}
rtc_static_library("rtc_media_base") {
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# Enabling GN check triggers cyclic dependency error:
# :rtc_media_base ->
# ../pc:rtc_pc_base ->
# :rtc_data ->
# :rtc_media_base
check_includes = false
defines = []
libs = []
deps = []
public_deps = []
sources = [
"base/adaptedvideotracksource.cc",
"base/adaptedvideotracksource.h",
"base/audiosource.h",
"base/codec.cc",
"base/codec.h",
"base/cryptoparams.h",
"base/device.h",
"base/mediachannel.h",
"base/mediaconstants.cc",
"base/mediaconstants.h",
"base/mediaengine.cc",
"base/mediaengine.h",
"base/rtpdataengine.cc",
"base/rtpdataengine.h",
"base/rtputils.cc",
"base/rtputils.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/videosourcebase.cc",
"base/videosourcebase.h",
# TODO(aleloi): add "base/videosinkinterface.h"
"base/videosourceinterface.cc",
# TODO(aleloi): add "base/videosourceinterface.h"
]
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" ]
}
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" ]
}
deps += [
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../p2p",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
]
public_deps += [ ":rtc_h264_profile_id" ]
if (is_nacl) {
deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
}
}
rtc_static_library("rtc_audio_video") {
defines = []
libs = []
deps = []
sources = [
"engine/adm_helpers.cc",
"engine/adm_helpers.h",
"engine/apm_helpers.cc",
"engine/apm_helpers.h",
"engine/internaldecoderfactory.cc",
"engine/internaldecoderfactory.h",
"engine/internalencoderfactory.cc",
"engine/internalencoderfactory.h",
"engine/nullwebrtcvideoengine.h",
"engine/payload_type_mapper.cc",
"engine/payload_type_mapper.h",
"engine/scopedvideoencoder.cc",
"engine/scopedvideoencoder.h",
"engine/simulcast.cc",
"engine/simulcast.h",
"engine/simulcast_encoder_adapter.cc",
"engine/simulcast_encoder_adapter.h",
"engine/videodecodersoftwarefallbackwrapper.cc",
"engine/videodecodersoftwarefallbackwrapper.h",
"engine/videoencodersoftwarefallbackwrapper.cc",
"engine/videoencodersoftwarefallbackwrapper.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/webrtcvideoengine.cc",
"engine/webrtcvideoengine.h",
"engine/webrtcvoe.h",
"engine/webrtcvoiceengine.cc",
"engine/webrtcvoiceengine.h",
]
configs += [ ":rtc_media_warnings_config" ]
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_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" ]
}
if (rtc_opus_support_120ms_ptime) {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
} else {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=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" ]
}
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 (rtc_enable_protobuf) {
deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
} else {
deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
}
deps += [
":rtc_media_base",
"..:webrtc_common",
"../api:call_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:transport_api",
"../api:video_frame_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:video_stream_api",
"../common_video:common_video",
"../modules/audio_coding:rent_a_codec",
"../modules/audio_device:audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing",
"../modules/audio_processing/aec_dump",
"../modules/video_capture:video_capture_module",
"../modules/video_coding",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
"../p2p:rtc_p2p",
"../pc:rtc_pc_base",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:sequenced_task_checker",
"../system_wrappers",
"../video",
"../voice_engine",
]
}
rtc_static_library("rtc_data") {
defines = []
deps = []
if (rtc_enable_sctp) {
sources = [
"sctp/sctptransport.cc",
"sctp/sctptransport.h",
"sctp/sctptransportinternal.h",
]
}
configs += [ ":rtc_media_warnings_config" ]
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_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_sctp && 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" ]
}
deps += [
":rtc_media_base",
"..:webrtc_common",
"../api:call_api",
"../api:transport_api",
"../p2p:rtc_p2p",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
]
}
rtc_source_set("rtc_media") {
public_deps = [
":rtc_audio_video",
":rtc_data",
]
}
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_media_tests_utils") {
testonly = true
include_dirs = []
public_deps = []
deps = [
"../call:video_stream_api",
"../modules/audio_coding:rent_a_codec",
"../modules/audio_processing:audio_processing",
"../modules/rtp_rtcp:rtp_rtcp",
"../p2p:rtc_p2p",
]
sources = [
"base/fakemediaengine.h",
"base/fakenetworkinterface.h",
"base/fakertp.cc",
"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 (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps += [
":rtc_media",
":rtc_media_base",
"..:webrtc_common",
"../api:call_api",
"../api:video_frame_api",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../test:test_support",
"//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",
]
}
if (!is_win) {
cflags = [ "-Wno-sign-compare" ]
}
}
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 = [
"../pc:rtc_pc",
"../test:field_trial",
]
sources = [
"base/codec_unittest.cc",
"base/rtpdataengine_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",
"engine/apm_helpers_unittest.cc",
"engine/internaldecoderfactory_unittest.cc",
"engine/nullwebrtcvideoengine_unittest.cc",
"engine/payload_type_mapper_unittest.cc",
"engine/simulcast_encoder_adapter_unittest.cc",
"engine/simulcast_unittest.cc",
"engine/videodecodersoftwarefallbackwrapper_unittest.cc",
"engine/videoencodersoftwarefallbackwrapper_unittest.cc",
"engine/webrtcmediaengine_unittest.cc",
"engine/webrtcvideocapturer_unittest.cc",
"engine/webrtcvideoencoderfactory_unittest.cc",
"engine/webrtcvideoengine_unittest.cc",
]
# TODO(kthelgason): Reenable this test on iOS.
# See bugs.webrtc.org/5569
if (!is_ios) {
sources += [ "engine/webrtcvoiceengine_unittest.cc" ]
}
if (rtc_enable_sctp) {
sources += [ "sctp/sctptransport_unittest.cc" ]
}
configs += [ ":rtc_media_unittests_config" ]
if (rtc_use_h264) {
defines += [ "WEBRTC_USE_H264" ]
}
if (rtc_opus_support_120ms_ptime) {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
} else {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
}
if (is_win) {
cflags = [
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
"/wd4373", # virtual function override.
"/wd4389", # signed/unsigned mismatch.
]
}
if (!build_with_chromium && is_clang) {
suppressed_configs += [
"//build/config/clang:extra_warnings",
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
"//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 += [
":rtc_media",
":rtc_media_base",
":rtc_media_tests_utils",
"../api:video_frame_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../audio",
"../call:call_interfaces",
"../common_video:common_video",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_processing:audio_processing",
"../modules/video_coding:simulcast_test_utility",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_vp8",
"../p2p:p2p_test_utils",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
"../system_wrappers:metrics_default",
"../test:audio_codec_mocks",
"../test:test_support",
"../voice_engine:voice_engine",
]
}
}