modules/rtp_rtcp/BUILD.gn

# Copyright (c) 2014 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")

rtc_source_set("rtp_rtcp_format") {
  public = [
    "include/rtp_cvo.h",
    "include/rtp_header_extension_map.h",
    "include/rtp_rtcp_defines.h",
    "source/byte_io.h",
    "source/rtcp_packet.h",
    "source/rtcp_packet/app.h",
    "source/rtcp_packet/bye.h",
    "source/rtcp_packet/common_header.h",
    "source/rtcp_packet/compound_packet.h",
    "source/rtcp_packet/dlrr.h",
    "source/rtcp_packet/extended_jitter_report.h",
    "source/rtcp_packet/extended_reports.h",
    "source/rtcp_packet/fir.h",
    "source/rtcp_packet/nack.h",
    "source/rtcp_packet/pli.h",
    "source/rtcp_packet/psfb.h",
    "source/rtcp_packet/rapid_resync_request.h",
    "source/rtcp_packet/receiver_report.h",
    "source/rtcp_packet/remb.h",
    "source/rtcp_packet/report_block.h",
    "source/rtcp_packet/rrtr.h",
    "source/rtcp_packet/rtpfb.h",
    "source/rtcp_packet/sdes.h",
    "source/rtcp_packet/sender_report.h",
    "source/rtcp_packet/target_bitrate.h",
    "source/rtcp_packet/tmmb_item.h",
    "source/rtcp_packet/tmmbn.h",
    "source/rtcp_packet/tmmbr.h",
    "source/rtcp_packet/transport_feedback.h",
    "source/rtcp_packet/voip_metric.h",
    "source/rtp_header_extensions.h",
    "source/rtp_packet.h",
    "source/rtp_packet_received.h",
    "source/rtp_packet_to_send.h",
  ]
  sources = [
    "include/rtp_rtcp_defines.cc",
    "source/rtcp_packet.cc",
    "source/rtcp_packet/app.cc",
    "source/rtcp_packet/bye.cc",
    "source/rtcp_packet/common_header.cc",
    "source/rtcp_packet/compound_packet.cc",
    "source/rtcp_packet/dlrr.cc",
    "source/rtcp_packet/extended_jitter_report.cc",
    "source/rtcp_packet/extended_reports.cc",
    "source/rtcp_packet/fir.cc",
    "source/rtcp_packet/nack.cc",
    "source/rtcp_packet/pli.cc",
    "source/rtcp_packet/psfb.cc",
    "source/rtcp_packet/rapid_resync_request.cc",
    "source/rtcp_packet/receiver_report.cc",
    "source/rtcp_packet/remb.cc",
    "source/rtcp_packet/report_block.cc",
    "source/rtcp_packet/rrtr.cc",
    "source/rtcp_packet/rtpfb.cc",
    "source/rtcp_packet/sdes.cc",
    "source/rtcp_packet/sender_report.cc",
    "source/rtcp_packet/target_bitrate.cc",
    "source/rtcp_packet/tmmb_item.cc",
    "source/rtcp_packet/tmmbn.cc",
    "source/rtcp_packet/tmmbr.cc",
    "source/rtcp_packet/transport_feedback.cc",
    "source/rtcp_packet/voip_metric.cc",
    "source/rtp_header_extension_map.cc",
    "source/rtp_header_extensions.cc",
    "source/rtp_packet.cc",
    "source/rtp_packet_received.cc",
  ]

  deps = [
    "..:module_api",
    "../..:webrtc_common",
    "../../api:array_view",
    "../../api:libjingle_peerconnection_api",
    "../../api:optional",
    "../../api/audio_codecs:audio_codecs_api",
    "../../common_video",
    "../../rtc_base:rtc_base_approved",
    "../../system_wrappers",
  ]
}

rtc_static_library("rtp_rtcp") {
  sources = [
    "include/flexfec_receiver.h",
    "include/flexfec_sender.h",
    "include/receive_statistics.h",
    "include/remote_ntp_time_estimator.h",
    "include/rtp_header_parser.h",
    "include/rtp_payload_registry.h",
    "include/rtp_receiver.h",
    "include/rtp_rtcp.h",
    "include/ulpfec_receiver.h",
    "source/dtmf_queue.cc",
    "source/dtmf_queue.h",
    "source/fec_private_tables_bursty.h",
    "source/fec_private_tables_random.h",
    "source/flexfec_header_reader_writer.cc",
    "source/flexfec_header_reader_writer.h",
    "source/flexfec_receiver.cc",
    "source/flexfec_sender.cc",
    "source/forward_error_correction.cc",
    "source/forward_error_correction.h",
    "source/forward_error_correction_internal.cc",
    "source/forward_error_correction_internal.h",
    "source/packet_loss_stats.cc",
    "source/packet_loss_stats.h",
    "source/playout_delay_oracle.cc",
    "source/playout_delay_oracle.h",
    "source/receive_statistics_impl.cc",
    "source/receive_statistics_impl.h",
    "source/remote_ntp_time_estimator.cc",
    "source/rtcp_nack_stats.cc",
    "source/rtcp_nack_stats.h",
    "source/rtcp_receiver.cc",
    "source/rtcp_receiver.h",
    "source/rtcp_sender.cc",
    "source/rtcp_sender.h",
    "source/rtp_format.cc",
    "source/rtp_format.h",
    "source/rtp_format_h264.cc",
    "source/rtp_format_h264.h",
    "source/rtp_format_video_generic.cc",
    "source/rtp_format_video_generic.h",
    "source/rtp_format_vp8.cc",
    "source/rtp_format_vp8.h",
    "source/rtp_format_vp9.cc",
    "source/rtp_format_vp9.h",
    "source/rtp_header_parser.cc",
    "source/rtp_packet_history.cc",
    "source/rtp_packet_history.h",
    "source/rtp_payload_registry.cc",
    "source/rtp_receiver_audio.cc",
    "source/rtp_receiver_audio.h",
    "source/rtp_receiver_impl.cc",
    "source/rtp_receiver_impl.h",
    "source/rtp_receiver_strategy.cc",
    "source/rtp_receiver_strategy.h",
    "source/rtp_receiver_video.cc",
    "source/rtp_receiver_video.h",
    "source/rtp_rtcp_config.h",
    "source/rtp_rtcp_impl.cc",
    "source/rtp_rtcp_impl.h",
    "source/rtp_sender.cc",
    "source/rtp_sender.h",
    "source/rtp_sender_audio.cc",
    "source/rtp_sender_audio.h",
    "source/rtp_sender_video.cc",
    "source/rtp_sender_video.h",
    "source/rtp_utility.cc",
    "source/rtp_utility.h",
    "source/time_util.cc",
    "source/time_util.h",
    "source/tmmbr_help.cc",
    "source/tmmbr_help.h",
    "source/ulpfec_generator.cc",
    "source/ulpfec_generator.h",
    "source/ulpfec_header_reader_writer.cc",
    "source/ulpfec_header_reader_writer.h",
    "source/ulpfec_receiver_impl.cc",
    "source/ulpfec_receiver_impl.h",
    "source/video_codec_information.h",
  ]

  if (rtc_enable_bwe_test_logging) {
    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
  } else {
    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
  }

  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 = [
    "..:module_api",
    "../..:webrtc_common",
    "../../api:array_view",
    "../../api:libjingle_peerconnection_api",
    "../../api:optional",
    "../../api:transport_api",
    "../../api/audio_codecs:audio_codecs_api",
    "../../common_video",
    "../../logging:rtc_event_log_api",
    "../../rtc_base:gtest_prod",
    "../../rtc_base:rtc_base_approved",
    "../../rtc_base:sequenced_task_checker",
    "../../system_wrappers",
    "../audio_coding:audio_format_conversion",
    "../remote_bitrate_estimator",
  ]

  public_deps = [
    ":rtp_rtcp_format",
  ]

  # TODO(jschuh): Bug 1348: fix this warning.
  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]

  if (is_win) {
    cflags = [
      # TODO(kjellander): Bug 261: fix this warning.
      "/wd4373",  # virtual function override.
    ]
  }
}

rtc_source_set("rtcp_transceiver") {
  public = [
    "source/rtcp_transceiver_config.h",
    "source/rtcp_transceiver_impl.h",
  ]
  sources = [
    "source/rtcp_transceiver_config.cc",
    "source/rtcp_transceiver_impl.cc",
  ]
  deps = [
    ":rtp_rtcp",
    "../../api:array_view",
    "../../api:transport_api",
    "../../rtc_base:rtc_base_approved",
  ]
}

rtc_source_set("fec_test_helper") {
  testonly = true
  sources = [
    "source/fec_test_helper.cc",
    "source/fec_test_helper.h",
  ]
  deps = [
    ":rtp_rtcp",
    "..:module_api",
    "../../rtc_base:rtc_base_approved",
  ]

  # TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
  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("mock_rtp_rtcp") {
  testonly = true
  sources = [
    "mocks/mock_recovered_packet_receiver.h",
    "mocks/mock_rtcp_rtt_stats.h",
    "mocks/mock_rtp_rtcp.h",
  ]
  deps = [
    ":rtp_rtcp",
    "..:module_api",
    "../../api:optional",
    "../../rtc_base:rtc_base_approved",
    "../../test:test_support",
  ]
}

if (rtc_include_tests) {
  rtc_executable("test_packet_masks_metrics") {
    testonly = true

    sources = [
      "test/testFec/average_residual_loss_xor_codes.h",
      "test/testFec/test_packet_masks_metrics.cc",
    ]

    deps = [
      ":rtp_rtcp",
      "../../test:test_main",
      "//testing/gtest",
    ]
  }  # test_packet_masks_metrics

  rtc_source_set("rtp_rtcp_modules_tests") {
    testonly = true

    # Skip restricting visibility on mobile platforms since the tests on those
    # gets additional generated targets which would require many lines here to
    # cover (which would be confusing to read and hard to maintain).
    if (!is_android && !is_ios) {
      visibility = [ "../../modules:modules_tests" ]
    }
    sources = [
      "test/testFec/test_fec.cc",
    ]
    deps = [
      ":rtp_rtcp",
      "../../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("rtp_rtcp_unittests") {
    testonly = true

    # Skip restricting visibility on mobile platforms since the tests on those
    # gets additional generated targets which would require many lines here to
    # cover (which would be confusing to read and hard to maintain).
    if (!is_android && !is_ios) {
      visibility = [ "..:modules_unittests" ]
    }
    sources = [
      "source/byte_io_unittest.cc",
      "source/flexfec_header_reader_writer_unittest.cc",
      "source/flexfec_receiver_unittest.cc",
      "source/flexfec_sender_unittest.cc",
      "source/nack_rtx_unittest.cc",
      "source/packet_loss_stats_unittest.cc",
      "source/playout_delay_oracle_unittest.cc",
      "source/receive_statistics_unittest.cc",
      "source/remote_ntp_time_estimator_unittest.cc",
      "source/rtcp_nack_stats_unittest.cc",
      "source/rtcp_packet/app_unittest.cc",
      "source/rtcp_packet/bye_unittest.cc",
      "source/rtcp_packet/common_header_unittest.cc",
      "source/rtcp_packet/compound_packet_unittest.cc",
      "source/rtcp_packet/dlrr_unittest.cc",
      "source/rtcp_packet/extended_jitter_report_unittest.cc",
      "source/rtcp_packet/extended_reports_unittest.cc",
      "source/rtcp_packet/fir_unittest.cc",
      "source/rtcp_packet/nack_unittest.cc",
      "source/rtcp_packet/pli_unittest.cc",
      "source/rtcp_packet/rapid_resync_request_unittest.cc",
      "source/rtcp_packet/receiver_report_unittest.cc",
      "source/rtcp_packet/remb_unittest.cc",
      "source/rtcp_packet/report_block_unittest.cc",
      "source/rtcp_packet/rrtr_unittest.cc",
      "source/rtcp_packet/sdes_unittest.cc",
      "source/rtcp_packet/sender_report_unittest.cc",
      "source/rtcp_packet/target_bitrate_unittest.cc",
      "source/rtcp_packet/tmmbn_unittest.cc",
      "source/rtcp_packet/tmmbr_unittest.cc",
      "source/rtcp_packet/transport_feedback_unittest.cc",
      "source/rtcp_packet/voip_metric_unittest.cc",
      "source/rtcp_packet_unittest.cc",
      "source/rtcp_receiver_unittest.cc",
      "source/rtcp_sender_unittest.cc",
      "source/rtcp_transceiver_impl_unittest.cc",
      "source/rtp_fec_unittest.cc",
      "source/rtp_format_h264_unittest.cc",
      "source/rtp_format_video_generic_unittest.cc",
      "source/rtp_format_vp8_test_helper.cc",
      "source/rtp_format_vp8_test_helper.h",
      "source/rtp_format_vp8_unittest.cc",
      "source/rtp_format_vp9_unittest.cc",
      "source/rtp_header_extension_map_unittest.cc",
      "source/rtp_packet_history_unittest.cc",
      "source/rtp_packet_unittest.cc",
      "source/rtp_payload_registry_unittest.cc",
      "source/rtp_receiver_unittest.cc",
      "source/rtp_rtcp_impl_unittest.cc",
      "source/rtp_sender_unittest.cc",
      "source/rtp_utility_unittest.cc",
      "source/time_util_unittest.cc",
      "source/ulpfec_generator_unittest.cc",
      "source/ulpfec_header_reader_writer_unittest.cc",
      "source/ulpfec_receiver_unittest.cc",
      "test/testAPI/test_api.cc",
      "test/testAPI/test_api.h",
      "test/testAPI/test_api_audio.cc",
      "test/testAPI/test_api_rtcp.cc",
      "test/testAPI/test_api_video.cc",
    ]
    deps = [
      ":fec_test_helper",
      ":mock_rtp_rtcp",
      ":rtcp_transceiver",
      ":rtp_rtcp",
      "..:module_api",
      "../..:webrtc_common",
      "../../api:array_view",
      "../../api:libjingle_peerconnection_api",
      "../../api:transport_api",
      "../../call:rtp_receiver",
      "../../common_video:common_video",
      "../../logging:rtc_event_log_api",
      "../../rtc_base:rtc_base_approved",
      "../../system_wrappers:system_wrappers",
      "../../test:field_trial",
      "../../test:rtp_test_utils",
      "../../test:test_common",
      "../../test:test_support",
      "../audio_coding:audio_format_conversion",
      "//testing/gmock",
    ]

    # TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
    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" ]
    }
  }
}