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webrtc_m130/pc/BUILD.gn
Karl Wiberg bb23c838f5 GN hack to tag targets as poisonous (and use it with audio codecs) 
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.

This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).

Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-23 13:41:47 +00:00

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# 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("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
group("pc") {
deps = [
":rtc_pc",
]
}
config("rtc_pc_config") {
defines = []
if (rtc_enable_sctp) {
defines += [ "HAVE_SCTP" ]
}
}
rtc_static_library("rtc_pc_base") {
visibility = [ "*" ]
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
defines = []
sources = [
"audiomonitor.h",
"channel.cc",
"channel.h",
"channelmanager.cc",
"channelmanager.h",
"currentspeakermonitor.cc",
"currentspeakermonitor.h",
"dtlssrtptransport.cc",
"dtlssrtptransport.h",
"externalhmac.cc",
"externalhmac.h",
"jseptransport.cc",
"jseptransport.h",
"jseptransportcontroller.cc",
"jseptransportcontroller.h",
"mediasession.cc",
"mediasession.h",
"rtcpmuxfilter.cc",
"rtcpmuxfilter.h",
"rtpmediautils.cc",
"rtpmediautils.h",
"rtptransport.cc",
"rtptransport.h",
"rtptransportinternal.h",
"rtptransportinternaladapter.h",
"sessiondescription.cc",
"sessiondescription.h",
"srtpfilter.cc",
"srtpfilter.h",
"srtpsession.cc",
"srtpsession.h",
"srtptransport.cc",
"srtptransport.h",
"transportstats.cc",
"transportstats.h",
]
deps = [
"..:webrtc_common",
"../api:array_view",
"../api:call_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:ortc_api",
"../api:video_frame_api",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../common_video:common_video",
"../media:rtc_data",
"../media:rtc_h264_profile_id",
"../media:rtc_media_base",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue",
"../rtc_base:stringutils",
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
public_configs = [ ":rtc_pc_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" ]
}
}
rtc_source_set("rtc_pc") {
visibility = [ "*" ]
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
deps = [
":rtc_pc_base",
"../media:rtc_audio_video",
]
}
config("libjingle_peerconnection_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 && !is_clang) {
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
}
}
rtc_static_library("peerconnection") {
visibility = [ "*" ]
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
cflags = []
sources = [
"audiotrack.cc",
"audiotrack.h",
"datachannel.cc",
"datachannel.h",
"dtmfsender.cc",
"dtmfsender.h",
"iceserverparsing.cc",
"iceserverparsing.h",
"jsepicecandidate.cc",
"jsepsessiondescription.cc",
"localaudiosource.cc",
"localaudiosource.h",
"mediastream.cc",
"mediastream.h",
"mediastreamobserver.cc",
"mediastreamobserver.h",
"mediastreamtrack.h",
"peerconnection.cc",
"peerconnection.h",
"peerconnectionfactory.cc",
"peerconnectionfactory.h",
"peerconnectioninternal.h",
"remoteaudiosource.cc",
"remoteaudiosource.h",
"rtcstatscollector.cc",
"rtcstatscollector.h",
"rtcstatstraversal.cc",
"rtcstatstraversal.h",
"rtpreceiver.cc",
"rtpreceiver.h",
"rtpsender.cc",
"rtpsender.h",
"rtptransceiver.cc",
"rtptransceiver.h",
"sctputils.cc",
"sctputils.h",
"sdputils.cc",
"sdputils.h",
"statscollector.cc",
"statscollector.h",
"streamcollection.h",
"trackmediainfomap.cc",
"trackmediainfomap.h",
"videocapturertracksource.cc",
"videocapturertracksource.h",
"videotrack.cc",
"videotrack.h",
"videotracksource.cc",
"videotracksource.h",
"webrtcsdp.cc",
"webrtcsdp.h",
"webrtcsessiondescriptionfactory.cc",
"webrtcsessiondescriptionfactory.h",
]
configs += [ ":libjingle_peerconnection_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" ]
}
deps = [
":rtc_pc_base",
"..:webrtc_common",
"../api:call_api",
"../api:fec_controller_api",
"../api:libjingle_peerconnection_api",
"../api:optional",
"../api:rtc_stats_api",
"../api:video_frame_api",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video:common_video",
"../logging:ice_log",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_output",
"../media:rtc_data",
"../media:rtc_media_base",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../stats",
"../system_wrappers",
"../system_wrappers:field_trial_api",
]
}
# This target implements CreatePeerConnectionFactory methods that will create a
# PeerConnection will full functionality (audio, video and data). Applications
# that wish to reduce their binary size by ommitting functionality they don't
# need should use CreateModularCreatePeerConnectionFactory instead, using the
# "peerconnection" build target and other targets specific to their
# requrements. See comment in peerconnectionfactoryinterface.h.
rtc_static_library("create_pc_factory") {
sources = [
"createpeerconnectionfactory.cc",
]
deps = [
"../api:callfactory_api",
"../api:libjingle_peerconnection_api",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../modules/audio_device:audio_device",
"../modules/audio_processing:audio_processing",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
]
configs += [ ":libjingle_peerconnection_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" ]
}
}
rtc_source_set("libjingle_peerconnection") {
visibility = [ "*" ]
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
deps = [
":create_pc_factory",
":peerconnection",
"../api:libjingle_peerconnection_api",
]
}
if (rtc_include_tests) {
config("rtc_pc_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.
if (!is_win && !is_clang) {
cflags = [ "-Wno-maybe-uninitialized" ] # Only exists for GCC.
}
}
rtc_test("rtc_pc_unittests") {
testonly = true
sources = [
"channel_unittest.cc",
"channelmanager_unittest.cc",
"currentspeakermonitor_unittest.cc",
"dtlssrtptransport_unittest.cc",
"jseptransport_unittest.cc",
"jseptransportcontroller_unittest.cc",
"mediasession_unittest.cc",
"rtcpmuxfilter_unittest.cc",
"rtptransport_unittest.cc",
"rtptransporttestutil.h",
"srtpfilter_unittest.cc",
"srtpsession_unittest.cc",
"srtptestutil.h",
"srtptransport_unittest.cc",
]
include_dirs = [ "//third_party/libsrtp/srtp" ]
configs += [ ":rtc_pc_unittests_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) {
libs = [ "strmiids.lib" ]
}
deps = [
":libjingle_peerconnection",
":pc_test_utils",
":rtc_pc",
":rtc_pc_base",
"../api:array_view",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../call:rtp_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../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",
"../system_wrappers:runtime_enabled_features_default",
"../test:test_support",
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
}
}
rtc_source_set("pc_test_utils") {
testonly = true
sources = [
"test/fakeaudiocapturemodule.cc",
"test/fakeaudiocapturemodule.h",
"test/fakedatachannelprovider.h",
"test/fakepeerconnectionbase.h",
"test/fakepeerconnectionforstats.h",
"test/fakeperiodicvideocapturer.h",
"test/fakeperiodicvideosource.h",
"test/fakertccertificategenerator.h",
"test/fakesctptransport.h",
"test/fakevideotrackrenderer.h",
"test/fakevideotracksource.h",
"test/mock_datachannel.h",
"test/mock_peerconnection.h",
"test/mock_rtpreceiverinternal.h",
"test/mock_rtpsenderinternal.h",
"test/mockpeerconnectionobservers.h",
"test/peerconnectiontestwrapper.cc",
"test/peerconnectiontestwrapper.h",
"test/rtcstatsobtainer.h",
"test/testsdpstrings.h",
]
deps = [
":libjingle_peerconnection",
":peerconnection",
":rtc_pc_base",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_data",
"../media:rtc_media",
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
"../modules/audio_device:audio_device",
"../p2p:p2p_test_utils",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
"../test:test_support",
]
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" ]
}
}
config("peerconnection_unittests_config") {
# The warnings below are enabled by default. Since GN orders compiler flags
# for a target before flags from configs, the only way to disable such
# warnings is by having them in a separate config, loaded from the target.
# 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=6267
# for -Wno-sign-compare
"-Wno-sign-compare",
]
}
if (!is_win) {
cflags = [ "-Wno-sign-compare" ]
}
}
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
"datachannel_unittest.cc",
"dtmfsender_unittest.cc",
"iceserverparsing_unittest.cc",
"jsepsessiondescription_unittest.cc",
"localaudiosource_unittest.cc",
"mediaconstraintsinterface_unittest.cc",
"mediastream_unittest.cc",
"peerconnection_bundle_unittest.cc",
"peerconnection_crypto_unittest.cc",
"peerconnection_datachannel_unittest.cc",
"peerconnection_ice_unittest.cc",
"peerconnection_integrationtest.cc",
"peerconnection_jsep_unittest.cc",
"peerconnection_media_unittest.cc",
"peerconnection_rtp_unittest.cc",
"peerconnection_signaling_unittest.cc",
"peerconnectionendtoend_unittest.cc",
"peerconnectionfactory_unittest.cc",
"peerconnectioninterface_unittest.cc",
"peerconnectionwrapper.cc",
"peerconnectionwrapper.h",
"proxy_unittest.cc",
"rtcstats_integrationtest.cc",
"rtcstatscollector_unittest.cc",
"rtcstatstraversal_unittest.cc",
"rtpmediautils_unittest.cc",
"rtpsenderreceiver_unittest.cc",
"sctputils_unittest.cc",
"statscollector_unittest.cc",
"test/fakeaudiocapturemodule_unittest.cc",
"test/testsdpstrings.h",
"trackmediainfomap_unittest.cc",
"videocapturertracksource_unittest.cc",
"videotrack_unittest.cc",
"webrtcsdp_unittest.cc",
]
if (rtc_enable_sctp) {
defines = [ "HAVE_SCTP" ]
}
configs += [ ":peerconnection_unittests_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" ]
}
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
if (is_win) {
cflags = [
"/wd4245", # conversion from int to size_t, signed/unsigned mismatch.
"/wd4389", # signed/unsigned mismatch.
]
}
deps = [
":peerconnection",
":rtc_pc_base",
"../api:libjingle_peerconnection_api",
"../api:mock_rtp",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../test:fileutils",
]
if (is_android) {
deps += [ ":android_black_magic" ]
}
deps += [
":libjingle_peerconnection",
":pc_test_utils",
"..:webrtc_common",
"../api:callfactory_api",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_test_api",
"../api:optional",
"../api:rtc_stats_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/audio_codecs/L16:audio_decoder_L16",
"../api/audio_codecs/L16:audio_encoder_L16",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../logging:rtc_event_log_impl_output",
"../media:rtc_audio_video",
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
"../modules/audio_processing:audio_processing",
"../modules/utility:utility",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../pc:rtc_pc",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_main",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
"../system_wrappers:metrics_default",
"../system_wrappers:runtime_enabled_features_default",
"../test:audio_codec_mocks",
"../test:test_support",
]
if (is_android) {
deps += [
"//testing/android/native_test:native_test_support",
# We need to depend on this one directly, or classloads will fail for
# the voice engine BuildInfo, for instance.
"../sdk/android:libjingle_peerconnection_java",
]
shard_timeout = 900
}
}
if (is_android) {
rtc_source_set("android_black_magic") {
# The android code uses hacky includes to chromium-base and the ssl code;
# having this in a separate target enables us to keep the peerconnection
# unit tests clean.
check_includes = false
testonly = true
sources = [
"test/androidtestinitializer.cc",
"test/androidtestinitializer.h",
]
deps = [
"../sdk/android:libjingle_peerconnection_jni",
"//testing/android/native_test:native_test_support",
]
}
}
}
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