webrtc/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("../../build/webrtc.gni")

source_set("rtp_rtcp") {
  sources = [
    # Common
    "interface/fec_receiver.h",
    "interface/receive_statistics.h",
    "interface/remote_ntp_time_estimator.h",
    "interface/rtp_header_parser.h",
    "interface/rtp_payload_registry.h",
    "interface/rtp_receiver.h",
    "interface/rtp_rtcp.h",
    "interface/rtp_rtcp_defines.h",
    "source/bitrate.cc",
    "source/bitrate.h",
    "source/byte_io.h",
    "source/fec_receiver_impl.cc",
    "source/fec_receiver_impl.h",
    "source/receive_statistics_impl.cc",
    "source/receive_statistics_impl.h",
    "source/remote_ntp_time_estimator.cc",
    "source/rtp_header_parser.cc",
    "source/rtp_rtcp_config.h",
    "source/rtp_rtcp_impl.cc",
    "source/rtp_rtcp_impl.h",
    "source/rtcp_packet.cc",
    "source/rtcp_packet.h",
    "source/rtcp_receiver.cc",
    "source/rtcp_receiver.h",
    "source/rtcp_receiver_help.cc",
    "source/rtcp_receiver_help.h",
    "source/rtcp_sender.cc",
    "source/rtcp_sender.h",
    "source/rtcp_utility.cc",
    "source/rtcp_utility.h",
    "source/rtp_header_extension.cc",
    "source/rtp_header_extension.h",
    "source/rtp_receiver_impl.cc",
    "source/rtp_receiver_impl.h",
    "source/rtp_sender.cc",
    "source/rtp_sender.h",
    "source/rtp_utility.cc",
    "source/rtp_utility.h",
    "source/ssrc_database.cc",
    "source/ssrc_database.h",
    "source/tmmbr_help.cc",
    "source/tmmbr_help.h",
    # Audio Files
    "source/dtmf_queue.cc",
    "source/dtmf_queue.h",
    "source/rtp_receiver_audio.cc",
    "source/rtp_receiver_audio.h",
    "source/rtp_sender_audio.cc",
    "source/rtp_sender_audio.h",
    # Video Files
    "source/fec_private_tables_random.h",
    "source/fec_private_tables_bursty.h",
    "source/forward_error_correction.cc",
    "source/forward_error_correction.h",
    "source/forward_error_correction_internal.cc",
    "source/forward_error_correction_internal.h",
    "source/producer_fec.cc",
    "source/producer_fec.h",
    "source/rtp_packet_history.cc",
    "source/rtp_packet_history.h",
    "source/rtp_payload_registry.cc",
    "source/rtp_receiver_strategy.cc",
    "source/rtp_receiver_strategy.h",
    "source/rtp_receiver_video.cc",
    "source/rtp_receiver_video.h",
    "source/rtp_sender_video.cc",
    "source/rtp_sender_video.h",
    "source/video_codec_information.h",
    "source/rtp_format.cc",
    "source/rtp_format.h",
    "source/rtp_format_h264.cc",
    "source/rtp_format_h264.h",
    "source/rtp_format_vp8.cc",
    "source/rtp_format_vp8.h",
    "source/rtp_format_video_generic.cc",
    "source/rtp_format_video_generic.h",
    "source/vp8_partition_aggregator.cc",
    "source/vp8_partition_aggregator.h",
    # Mocks
    "mocks/mock_rtp_rtcp.h",
    "source/mock/mock_rtp_payload_strategy.h",
  ]

  configs += [ "../..:common_config" ]
  public_configs = [ "../..:common_inherited_config" ]

  if (is_clang) {
    # Suppress warnings from Chrome's Clang plugins.
    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
    configs -= [ "//build/config/clang:find_bad_constructs" ]
  }

  deps = [
    "../..:webrtc_common",
    "../../system_wrappers",
    "../pacing",
    "../remote_bitrate_estimator",
  ]

  if (is_win) {
    cflags = [
      # TODO(jschuh): Bug 1348: fix this warning.
      "/wd4267",  # size_t to int truncations
      # TODO(kjellander): Bug 261: fix this warning.
      "/wd4373",  # virtual function override.
    ]
  }
}