| # 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 = [ |
| "channel.cc", |
| "channel.h", |
| "channelinterface.h", |
| "channelmanager.cc", |
| "channelmanager.h", |
| "dtlssrtptransport.cc", |
| "dtlssrtptransport.h", |
| "dtlstransport.cc", |
| "dtlstransport.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", |
| "simulcastdescription.cc", |
| "simulcastdescription.h", |
| "srtpfilter.cc", |
| "srtpfilter.h", |
| "srtpsession.cc", |
| "srtpsession.h", |
| "srtptransport.cc", |
| "srtptransport.h", |
| "transportstats.cc", |
| "transportstats.h", |
| ] |
| |
| deps = [ |
| "..:webrtc_common", |
| "../api:array_view", |
| "../api:audio_options_api", |
| "../api:call_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:ortc_api", |
| "../api/video:video_frame", |
| "../call:call_interfaces", |
| "../call:rtp_interfaces", |
| "../call:rtp_receiver", |
| "../common_video:common_video", |
| "../logging:rtc_event_log_api", |
| "../media:rtc_data", |
| "../media:rtc_h264_profile_id", |
| "../media:rtc_media_base", |
| "../media:rtc_media_config", |
| "../modules/rtp_rtcp:rtp_rtcp_format", |
| "../p2p:rtc_p2p", |
| "../rtc_base:checks", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base/third_party/base64", |
| "../rtc_base/third_party/sigslot", |
| "../system_wrappers:metrics", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| 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. |
| "software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove. |
| ] |
| deps = [ |
| ":rtc_pc_base", |
| "../media:rtc_audio_video", |
| ] |
| } |
| |
| 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", |
| "rtcstatstraversal.cc", |
| "rtcstatstraversal.h", |
| "rtpparametersconversion.cc", |
| "rtpparametersconversion.h", |
| "rtpreceiver.cc", |
| "rtpreceiver.h", |
| "rtpsender.cc", |
| "rtpsender.h", |
| "rtptransceiver.cc", |
| "rtptransceiver.h", |
| "sctputils.cc", |
| "sctputils.h", |
| "sdpserializer.cc", |
| "sdpserializer.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", |
| ] |
| |
| 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:array_view", |
| "../api:audio_options_api", |
| "../api:call_api", |
| "../api:fec_controller_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:rtc_stats_api", |
| "../api/video:video_frame", |
| "../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/system:rtc_export", |
| "../rtc_base/third_party/base64", |
| "../rtc_base/third_party/sigslot", |
| "../stats", |
| "../system_wrappers", |
| "../system_wrappers:field_trial", |
| "../system_wrappers:metrics", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| rtc_source_set("libjingle_peerconnection") { |
| visibility = [ "*" ] |
| allow_poison = [ |
| "audio_codecs", # TODO(bugs.webrtc.org/8396): Remove. |
| "software_video_codecs", # TODO(bugs.webrtc.org/7925): Remove. |
| ] |
| deps = [ |
| ":peerconnection", |
| "../api:create_peerconnection_factory", |
| "../api:libjingle_peerconnection_api", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_test("rtc_pc_unittests") { |
| testonly = true |
| |
| sources = [ |
| "channel_unittest.cc", |
| "channelmanager_unittest.cc", |
| "dtlssrtptransport_unittest.cc", |
| "jseptransport_unittest.cc", |
| "jseptransportcontroller_unittest.cc", |
| "mediasession_unittest.cc", |
| "rtcpmuxfilter_unittest.cc", |
| "rtptransport_unittest.cc", |
| "rtptransporttestutil.h", |
| "sessiondescription_unittest.cc", |
| "srtpfilter_unittest.cc", |
| "srtpsession_unittest.cc", |
| "srtptestutil.h", |
| "srtptransport_unittest.cc", |
| ] |
| |
| include_dirs = [ "//third_party/libsrtp/srtp" ] |
| |
| 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:audio_options_api", |
| "../api:fake_media_transport", |
| "../api:libjingle_peerconnection_api", |
| "../call:rtp_interfaces", |
| "../call:rtp_receiver", |
| "../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:gunit_helpers", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_base_tests_main", |
| "../rtc_base:rtc_base_tests_utils", |
| "../rtc_base/third_party/sigslot", |
| "../system_wrappers:metrics", |
| "../test:test_support", |
| "//third_party/abseil-cpp/absl/memory", |
| ] |
| |
| if (rtc_build_libsrtp) { |
| deps += [ "//third_party/libsrtp" ] |
| } |
| |
| if (is_android) { |
| deps += [ "//testing/android/native_test:native_test_support" ] |
| } |
| } |
| |
| rtc_source_set("peerconnection_perf_tests") { |
| testonly = true |
| sources = [ |
| "peerconnection_rampup_tests.cc", |
| ] |
| deps = [ |
| ":pc_test_utils", |
| ":peerconnection_wrapper", |
| "../api:audio_options_api", |
| "../api:create_peerconnection_factory", |
| "../api:libjingle_peerconnection_api", |
| "../api:rtc_stats_api", |
| "../api/audio:audio_mixer_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/audio_codecs:builtin_audio_decoder_factory", |
| "../api/audio_codecs:builtin_audio_encoder_factory", |
| "../api/video_codecs:builtin_video_decoder_factory", |
| "../api/video_codecs:builtin_video_encoder_factory", |
| "../api/video_codecs:video_codecs_api", |
| "../media:rtc_media_tests_utils", |
| "../modules/audio_device:audio_device_api", |
| "../modules/audio_processing:api", |
| "../p2p:p2p_test_utils", |
| "../p2p:rtc_p2p", |
| "../pc:peerconnection", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_tests_utils", |
| "../system_wrappers:system_wrappers", |
| "../test:perf_test", |
| "../test:test_support", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| 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("peerconnection_wrapper") { |
| testonly = true |
| sources = [ |
| "peerconnectionwrapper.cc", |
| "peerconnectionwrapper.h", |
| ] |
| deps = [ |
| ":pc_test_utils", |
| "../api:libjingle_peerconnection_api", |
| "../api:rtc_stats_api", |
| "../pc:peerconnection", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base_approved", |
| "../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" ] |
| } |
| } |
| |
| 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/fakeperiodicvideosource.h", |
| "test/fakeperiodicvideotracksource.h", |
| "test/fakertccertificategenerator.h", |
| "test/fakesctptransport.h", |
| "test/fakevideotrackrenderer.h", |
| "test/fakevideotracksource.h", |
| "test/framegeneratorcapturervideotracksource.h", |
| "test/mock_channelinterface.h", |
| "test/mock_datachannel.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:audio_options_api", |
| "../api:create_peerconnection_factory", |
| "../api:libjingle_peerconnection_api", |
| "../api:libjingle_peerconnection_test_api", |
| "../api:rtc_stats_api", |
| "../api/audio:audio_mixer_api", |
| "../api/audio_codecs:audio_codecs_api", |
| "../api/video:video_frame", |
| "../api/video_codecs:builtin_video_decoder_factory", |
| "../api/video_codecs:builtin_video_encoder_factory", |
| "../api/video_codecs:video_codecs_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", |
| "../modules/audio_processing:api", |
| "../modules/audio_processing:audio_processing", |
| "../p2p:p2p_test_utils", |
| "../p2p:rtc_p2p", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base", |
| "../rtc_base:rtc_base_approved", |
| "../rtc_base:rtc_task_queue", |
| "../rtc_base/third_party/sigslot", |
| "../test:test_support", |
| "../test:video_test_common", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| 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_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_histogram_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", |
| "rtpparametersconversion_unittest.cc", |
| "rtpsenderreceiver_unittest.cc", |
| "rtptransceiver_unittest.cc", |
| "sctputils_unittest.cc", |
| "sdpserializer_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" ] |
| } |
| |
| 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 = [ |
| ":peerconnection", |
| ":rtc_pc_base", |
| "../api:array_view", |
| "../api:audio_options_api", |
| "../api:create_peerconnection_factory", |
| "../api:fake_frame_decryptor", |
| "../api:fake_frame_encryptor", |
| "../api:libjingle_logging_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:loopback_media_transport", |
| "../api:mock_rtp", |
| "../api/audio:audio_mixer_api", |
| "../api/units:time_delta", |
| "../logging:fake_rtc_event_log", |
| "../media:rtc_media_config", |
| "../modules/audio_device:audio_device_api", |
| "../modules/audio_processing:audio_processing_statistics", |
| "../rtc_base:checks", |
| "../rtc_base:gunit_helpers", |
| "../rtc_base:rtc_base_tests_utils", |
| "../rtc_base/third_party/base64", |
| "../rtc_base/third_party/sigslot:sigslot", |
| "../system_wrappers:metrics", |
| "../test:fileutils", |
| "//third_party/abseil-cpp/absl/memory", |
| "//third_party/abseil-cpp/absl/strings", |
| ] |
| if (is_android) { |
| deps += [ ":android_black_magic" ] |
| } |
| |
| deps += [ |
| ":libjingle_peerconnection", |
| ":pc_test_utils", |
| "..:webrtc_common", |
| "../api:callfactory_api", |
| "../api:fake_media_transport", |
| "../api:libjingle_peerconnection_test_api", |
| "../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", |
| "../api/video_codecs:builtin_video_decoder_factory", |
| "../api/video_codecs:builtin_video_encoder_factory", |
| "../api/video_codecs:video_codecs_api", |
| "../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:api", |
| "../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_task_queue", |
| "../rtc_base:safe_conversions", |
| "../test:audio_codec_mocks", |
| "../test:test_support", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| |
| 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", |
| ] |
| } |
| } |
| } |