| # 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 = [ "*" ] |
| defines = [] |
| sources = [ |
| "audiomonitor.h", |
| "bundlefilter.cc", |
| "bundlefilter.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", |
| "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", |
| "transportcontroller.cc", |
| "transportcontroller.h", |
| ] |
| |
| deps = [ |
| "..:webrtc_common", |
| "../api:call_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:optional", |
| "../api:ortc_api", |
| "../common_video:common_video", |
| "../media:rtc_data", |
| "../media:rtc_h264_profile_id", |
| "../media:rtc_media_base", |
| "../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 = [ "*" ] |
| 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 = [ "*" ] |
| 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", |
| "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:libjingle_peerconnection_api", |
| "../api:optional", |
| "../api:rtc_stats_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:audio_mixer_api", |
| "../api:callfactory_api", |
| "../api:libjingle_peerconnection_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 = [ "*" ] |
| 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 = [ |
| "bundlefilter_unittest.cc", |
| "channel_unittest.cc", |
| "channelmanager_unittest.cc", |
| "currentspeakermonitor_unittest.cc", |
| "dtlssrtptransport_unittest.cc", |
| "jseptransport_unittest.cc", |
| "mediasession_unittest.cc", |
| "rtcpmuxfilter_unittest.cc", |
| "rtptransport_unittest.cc", |
| "rtptransporttestutil.h", |
| "srtpfilter_unittest.cc", |
| "srtpsession_unittest.cc", |
| "srtptestutil.h", |
| "srtptransport_unittest.cc", |
| "transportcontroller_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:libjingle_peerconnection_api", |
| "../logging:rtc_event_log_api", |
| "../media:rtc_media_base", |
| "../media:rtc_media_tests_utils", |
| "../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/fakertccertificategenerator.h", |
| "test/fakesctptransport.h", |
| "test/faketransportcontroller.h", |
| "test/fakevideotrackrenderer.h", |
| "test/fakevideotracksource.h", |
| "test/mock_datachannel.h", |
| "test/mock_peerconnection.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_api", |
| "../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", |
| "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", |
| ] |
| 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_api", |
| "../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", |
| ] |
| } |
| } |
| } |
