# 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 = [ "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", "mocks/mock_rtp_rtcp.h", "source/bitrate.cc", "source/bitrate.h", "source/byte_io.h", "source/dtmf_queue.cc", "source/dtmf_queue.h", "source/fec_private_tables_bursty.h", "source/fec_private_tables_random.h", "source/fec_receiver_impl.cc", "source/fec_receiver_impl.h", "source/forward_error_correction.cc", "source/forward_error_correction.h", "source/forward_error_correction_internal.cc", "source/forward_error_correction_internal.h", "source/h264_bitstream_parser.cc", "source/h264_bitstream_parser.h", "source/h264_sps_parser.cc", "source/h264_sps_parser.h", "source/mock/mock_rtp_payload_strategy.h", "source/packet_loss_stats.cc", "source/packet_loss_stats.h", "source/producer_fec.cc", "source/producer_fec.h", "source/receive_statistics_impl.cc", "source/receive_statistics_impl.h", "source/remote_ntp_time_estimator.cc", "source/rtcp_packet.cc", "source/rtcp_packet.h", "source/rtcp_packet/transport_feedback.cc", "source/rtcp_packet/transport_feedback.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_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_extension.cc", "source/rtp_header_extension.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/ssrc_database.cc", "source/ssrc_database.h", "source/tmmbr_help.cc", "source/tmmbr_help.h", "source/video_codec_information.h", "source/vp8_partition_aggregator.cc", "source/vp8_partition_aggregator.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", "../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. ] } }