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diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
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+# Copyright (c) 2011 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.
+
+{
+ 'targets': [
+ {
+ 'target_name': 'rtp_rtcp',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
+ ],
+ '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/packet_loss_stats.cc',
+ 'source/packet_loss_stats.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_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_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/h264_bitstream_parser.cc',
+ 'source/h264_bitstream_parser.h',
+ 'source/h264_sps_parser.cc',
+ 'source/h264_sps_parser.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_vp9.cc',
+ 'source/rtp_format_vp9.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',
+ ], # source
+ # TODO(jschuh): Bug 1348: fix size_t to int truncations.
+ 'msvs_disabled_warnings': [ 4267, ],
+ },
+ ],
+}