pc/BUILD.gn

# 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",
    "channelinterface.h",
    "channelmanager.h",
    "dtls_srtp_transport.cc",
    "dtls_srtp_transport.h",
    "dtls_transport.cc",
    "dtls_transport.h",
    "dtlssrtptransport.h",
    "dtlstransport.h",
    "external_hmac.cc",
    "external_hmac.h",
    "externalhmac.h",
    "jsep_transport.cc",
    "jsep_transport.h",
    "jsep_transport_controller.cc",
    "jsep_transport_controller.h",
    "jseptransport.h",
    "jseptransportcontroller.h",
    "media_session.cc",
    "media_session.h",
    "mediasession.h",
    "rtcp_mux_filter.cc",
    "rtcp_mux_filter.h",
    "rtcpmuxfilter.h",
    "rtp_media_utils.cc",
    "rtp_media_utils.h",
    "rtp_transport.cc",
    "rtp_transport.h",
    "rtp_transport_internal.h",
    "rtp_transport_internal_adapter.h",
    "rtpmediautils.h",
    "rtptransport.h",
    "rtptransportinternal.h",
    "rtptransportinternaladapter.h",
    "session_description.cc",
    "session_description.h",
    "sessiondescription.h",
    "simulcast_description.cc",
    "simulcast_description.h",
    "simulcastdescription.h",
    "srtp_filter.cc",
    "srtp_filter.h",
    "srtp_session.cc",
    "srtp_session.h",
    "srtp_transport.cc",
    "srtp_transport.h",
    "srtpfilter.h",
    "srtpsession.h",
    "srtptransport.h",
    "transport_stats.cc",
    "transport_stats.h",
    "transportstats.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/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",
    "audiotrack.h",
    "data_channel.cc",
    "data_channel.h",
    "datachannel.h",
    "dtmf_sender.cc",
    "dtmf_sender.h",
    "dtmfsender.h",
    "ice_server_parsing.cc",
    "ice_server_parsing.h",
    "iceserverparsing.h",
    "jsep_ice_candidate.cc",
    "jsep_session_description.cc",
    "local_audio_source.cc",
    "local_audio_source.h",
    "localaudiosource.h",
    "media_stream.cc",
    "media_stream.h",
    "media_stream_observer.cc",
    "media_stream_observer.h",
    "media_stream_track.h",
    "mediastream.h",
    "mediastreamobserver.h",
    "mediastreamtrack.h",
    "peer_connection.cc",
    "peer_connection.h",
    "peer_connection_factory.cc",
    "peer_connection_factory.h",
    "peer_connection_internal.h",
    "peerconnection.h",
    "peerconnectionfactory.h",
    "peerconnectioninternal.h",
    "remote_audio_source.cc",
    "remote_audio_source.h",
    "remoteaudiosource.h",
    "rtc_stats_collector.cc",
    "rtc_stats_collector.h",
    "rtc_stats_traversal.cc",
    "rtc_stats_traversal.h",
    "rtcstatscollector.h",
    "rtcstatstraversal.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",
    "rtpparametersconversion.h",
    "rtpreceiver.h",
    "rtpsender.h",
    "rtptransceiver.h",
    "sctp_utils.cc",
    "sctp_utils.h",
    "sctputils.h",
    "sdp_serializer.cc",
    "sdp_serializer.h",
    "sdp_utils.cc",
    "sdp_utils.h",
    "sdpserializer.h",
    "sdputils.h",
    "stats_collector.cc",
    "stats_collector.h",
    "statscollector.h",
    "stream_collection.h",
    "streamcollection.h",
    "track_media_info_map.cc",
    "track_media_info_map.h",
    "trackmediainfomap.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",
    "videocapturertracksource.h",
    "videotrack.h",
    "videotracksource.h",
    "webrtc_sdp.cc",
    "webrtc_sdp.h",
    "webrtc_session_description_factory.cc",
    "webrtc_session_description_factory.h",
    "webrtcsdp.h",
    "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: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/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",
      "jsep_transport_controller_unittest.cc",
      "jsep_transport_unittest.cc",
      "media_session_unittest.cc",
      "rtcp_mux_filter_unittest.cc",
      "rtp_transport_unittest.cc",
      "rtptransporttestutil.h",
      "session_description_unittest.cc",
      "srtp_filter_unittest.cc",
      "srtp_session_unittest.cc",
      "srtp_transport_unittest.cc",
      "srtptestutil.h",
      "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: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 = [
      "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",
      "peerconnectionwrapper.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/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/frame_generator_capturer_video_track_source.h",
      "test/framegeneratorcapturervideotracksource.h",
      "test/mock_channel_interface.h",
      "test/mock_channelinterface.h",
      "test/mock_data_channel.h",
      "test/mock_datachannel.h",
      "test/mock_peer_connection_observers.h",
      "test/mock_rtp_receiver_internal.h",
      "test/mock_rtp_sender_internal.h",
      "test/mock_rtpreceiverinternal.h",
      "test/mock_rtpsenderinternal.h",
      "test/mockpeerconnectionobservers.h",
      "test/peer_connection_test_wrapper.cc",
      "test/peer_connection_test_wrapper.h",
      "test/peerconnectiontestwrapper.h",
      "test/rtc_stats_obtainer.h",
      "test/rtcstatsobtainer.h",
      "test/test_sdp_strings.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: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",
      "peerconnectionwrapper.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",
      "test/testsdpstrings.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/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",
        "test/androidtestinitializer.h",
      ]
      deps = [
        "../sdk/android:libjingle_peerconnection_jni",
        "//testing/android/native_test:native_test_support",
      ]
    }
  }
}