| # 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", |
| "channel_interface.h", |
| "channel_manager.cc", |
| "channel_manager.h", |
| "dtls_srtp_transport.cc", |
| "dtls_srtp_transport.h", |
| "dtls_transport.cc", |
| "dtls_transport.h", |
| "external_hmac.cc", |
| "external_hmac.h", |
| "ice_transport.cc", |
| "ice_transport.h", |
| "jsep_transport.cc", |
| "jsep_transport.h", |
| "jsep_transport_controller.cc", |
| "jsep_transport_controller.h", |
| "media_session.cc", |
| "media_session.h", |
| "rtcp_mux_filter.cc", |
| "rtcp_mux_filter.h", |
| "rtp_media_utils.cc", |
| "rtp_media_utils.h", |
| "rtp_transport.cc", |
| "rtp_transport.h", |
| "rtp_transport_internal.h", |
| "rtp_transport_internal_adapter.h", |
| "session_description.cc", |
| "session_description.h", |
| "simulcast_description.cc", |
| "simulcast_description.h", |
| "srtp_filter.cc", |
| "srtp_filter.h", |
| "srtp_session.cc", |
| "srtp_session.h", |
| "srtp_transport.cc", |
| "srtp_transport.h", |
| "transport_stats.cc", |
| "transport_stats.h", |
| ] |
| |
| deps = [ |
| "..:webrtc_common", |
| "../api:array_view", |
| "../api:audio_options_api", |
| "../api:call_api", |
| "../api:libjingle_peerconnection_api", |
| "../api:ortc_api", |
| "../api:scoped_refptr", |
| "../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:stringutils", |
| "../rtc_base/third_party/base64", |
| "../rtc_base/third_party/sigslot", |
| "../system_wrappers:metrics", |
| "//third_party/abseil-cpp/absl/algorithm:container", |
| "//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. |
| deps = [ |
| ":rtc_pc_base", |
| "../media:rtc_audio_video", |
| ] |
| } |
| |
| rtc_static_library("peerconnection") { |
| visibility = [ "*" ] |
| cflags = [] |
| sources = [ |
| "audio_track.cc", |
| "audio_track.h", |
| "data_channel.cc", |
| "data_channel.h", |
| "dtmf_sender.cc", |
| "dtmf_sender.h", |
| "ice_server_parsing.cc", |
| "ice_server_parsing.h", |
| "jsep_ice_candidate.cc", |
| "jsep_session_description.cc", |
| "local_audio_source.cc", |
| "local_audio_source.h", |
| "media_stream.cc", |
| "media_stream.h", |
| "media_stream_observer.cc", |
| "media_stream_observer.h", |
| "media_stream_track.h", |
| "peer_connection.cc", |
| "peer_connection.h", |
| "peer_connection_factory.cc", |
| "peer_connection_factory.h", |
| "peer_connection_internal.h", |
| "remote_audio_source.cc", |
| "remote_audio_source.h", |
| "rtc_stats_collector.cc", |
| "rtc_stats_collector.h", |
| "rtc_stats_traversal.cc", |
| "rtc_stats_traversal.h", |
| "rtp_parameters_conversion.cc", |
| "rtp_parameters_conversion.h", |
| "rtp_receiver.cc", |
| "rtp_receiver.h", |
| "rtp_sender.cc", |
| "rtp_sender.h", |
| "rtp_transceiver.cc", |
| "rtp_transceiver.h", |
| "sctp_utils.cc", |
| "sctp_utils.h", |
| "sdp_serializer.cc", |
| "sdp_serializer.h", |
| "sdp_utils.cc", |
| "sdp_utils.h", |
| "stats_collector.cc", |
| "stats_collector.h", |
| "stream_collection.h", |
| "track_media_info_map.cc", |
| "track_media_info_map.h", |
| "video_capturer_track_source.cc", |
| "video_capturer_track_source.h", |
| "video_track.cc", |
| "video_track.h", |
| "video_track_source.cc", |
| "video_track_source.h", |
| "webrtc_sdp.cc", |
| "webrtc_sdp.h", |
| "webrtc_session_description_factory.cc", |
| "webrtc_session_description_factory.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:scoped_refptr", |
| "../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/algorithm:container", |
| "//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. |
| deps = [ |
| ":peerconnection", |
| "../api:create_peerconnection_factory", |
| "../api:libjingle_peerconnection_api", |
| ] |
| } |
| |
| if (rtc_include_tests) { |
| rtc_test("rtc_pc_unittests") { |
| testonly = true |
| |
| sources = [ |
| "channel_manager_unittest.cc", |
| "channel_unittest.cc", |
| "dtls_srtp_transport_unittest.cc", |
| "dtlstransport_unittest.cc", |
| "ice_transport_unittest.cc", |
| "jsep_transport_controller_unittest.cc", |
| "jsep_transport_unittest.cc", |
| "media_session_unittest.cc", |
| "rtcp_mux_filter_unittest.cc", |
| "rtp_transport_unittest.cc", |
| "session_description_unittest.cc", |
| "srtp_filter_unittest.cc", |
| "srtp_session_unittest.cc", |
| "srtp_transport_unittest.cc", |
| "test/rtp_transport_test_util.h", |
| "test/srtp_test_util.h", |
| ] |
| |
| 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:ice_transport_factory", |
| "../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:fake_ice_transport", |
| "../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/algorithm:container", |
| "//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 = [ |
| "peer_connection_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:scoped_refptr", |
| "../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 = [ |
| "peer_connection_wrapper.cc", |
| "peer_connection_wrapper.h", |
| ] |
| deps = [ |
| ":pc_test_utils", |
| "../api:libjingle_peerconnection_api", |
| "../api:rtc_stats_api", |
| "../api:scoped_refptr", |
| "../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/fake_audio_capture_module.cc", |
| "test/fake_audio_capture_module.h", |
| "test/fake_data_channel_provider.h", |
| "test/fake_peer_connection_base.h", |
| "test/fake_peer_connection_for_stats.h", |
| "test/fake_periodic_video_source.h", |
| "test/fake_periodic_video_track_source.h", |
| "test/fake_rtc_certificate_generator.h", |
| "test/fake_sctp_transport.h", |
| "test/fake_video_track_renderer.h", |
| "test/fake_video_track_source.h", |
| "test/frame_generator_capturer_video_track_source.h", |
| "test/mock_channel_interface.h", |
| "test/mock_data_channel.h", |
| "test/mock_peer_connection_observers.h", |
| "test/mock_rtp_receiver_internal.h", |
| "test/mock_rtp_sender_internal.h", |
| "test/peer_connection_test_wrapper.cc", |
| "test/peer_connection_test_wrapper.h", |
| "test/rtc_stats_obtainer.h", |
| "test/test_sdp_strings.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:scoped_refptr", |
| "../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/task_utils:repeating_task", |
| "../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 = [ |
| "data_channel_unittest.cc", |
| "dtmf_sender_unittest.cc", |
| "ice_server_parsing_unittest.cc", |
| "jsep_session_description_unittest.cc", |
| "local_audio_source_unittest.cc", |
| "media_constraints_interface_unittest.cc", |
| "media_stream_unittest.cc", |
| "peer_connection_bundle_unittest.cc", |
| "peer_connection_crypto_unittest.cc", |
| "peer_connection_data_channel_unittest.cc", |
| "peer_connection_end_to_end_unittest.cc", |
| "peer_connection_factory_unittest.cc", |
| "peer_connection_histogram_unittest.cc", |
| "peer_connection_ice_unittest.cc", |
| "peer_connection_integrationtest.cc", |
| "peer_connection_interface_unittest.cc", |
| "peer_connection_jsep_unittest.cc", |
| "peer_connection_media_unittest.cc", |
| "peer_connection_rtp_unittest.cc", |
| "peer_connection_signaling_unittest.cc", |
| "peer_connection_wrapper.cc", |
| "peer_connection_wrapper.h", |
| "proxy_unittest.cc", |
| "rtc_stats_collector_unittest.cc", |
| "rtc_stats_integrationtest.cc", |
| "rtc_stats_traversal_unittest.cc", |
| "rtp_media_utils_unittest.cc", |
| "rtp_parameters_conversion_unittest.cc", |
| "rtp_sender_receiver_unittest.cc", |
| "rtp_transceiver_unittest.cc", |
| "sctp_utils_unittest.cc", |
| "sdp_serializer_unittest.cc", |
| "stats_collector_unittest.cc", |
| "test/fake_audio_capture_module_unittest.cc", |
| "test/test_sdp_strings.h", |
| "track_media_info_map_unittest.cc", |
| "video_capturer_track_source_unittest.cc", |
| "video_track_unittest.cc", |
| "webrtc_sdp_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:scoped_refptr", |
| "../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/algorithm:container", |
| "//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/android_test_initializer.cc", |
| "test/android_test_initializer.h", |
| ] |
| deps = [ |
| "../sdk/android:libjingle_peerconnection_jni", |
| "//testing/android/native_test:native_test_support", |
| ] |
| } |
| } |
| } |
