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# 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.
]
}
}
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