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| author | andresp@webrtc.org <andresp@webrtc.org> | 2015-02-25 11:50:17 +0000 |
|---|---|---|
| committer | andresp@webrtc.org <andresp@webrtc.org> | 2015-02-25 11:50:44 +0000 |
| commit | 749c60217d00a18544a72c7e9e4fe3b944dce9f2 (patch) | |
| tree | 154828ea7be715e8c056cacc389e886ef7b55684 /webrtc/modules/rtp_rtcp/rtp_rtcp.gypi | |
| parent | 5c928ebd1d5d61deca0b5458e31005ac5dc1c31f (diff) | |
| download | webrtc-749c60217d00a18544a72c7e9e4fe3b944dce9f2.tar.gz | |
Moved gypi to avoid presubmit warning about '..' when touching the files.
R=kjellander@webrtc.org,mflodman@webrtc.org
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39299004
Cr-Commit-Position: refs/heads/master@{#8503}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8503 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'webrtc/modules/rtp_rtcp/rtp_rtcp.gypi')
| -rw-r--r-- | webrtc/modules/rtp_rtcp/rtp_rtcp.gypi | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi new file mode 100644 index 0000000000..7a144e4ef5 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi @@ -0,0 +1,107 @@ +# 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:paced_sender', + '<(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/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', + ], # source + # TODO(jschuh): Bug 1348: fix size_t to int truncations. + 'msvs_disabled_warnings': [ 4267, ], + }, + ], +} |