f21ea918ad
This is needed to ensure we have the same build with GN as with GYP, since GYP includes the common.gypi on a global level. Several fixes has been needed in the past because some code have been built without the right defines. BUG=3441 R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/28589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
119 lines
3.7 KiB
Plaintext
119 lines
3.7 KiB
Plaintext
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../build/webrtc.gni")
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source_set("rtp_rtcp") {
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sources = [
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# Common
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"interface/fec_receiver.h",
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"interface/receive_statistics.h",
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"interface/remote_ntp_time_estimator.h",
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"interface/rtp_header_parser.h",
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"interface/rtp_payload_registry.h",
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"interface/rtp_receiver.h",
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"interface/rtp_rtcp.h",
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"interface/rtp_rtcp_defines.h",
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"source/bitrate.cc",
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"source/bitrate.h",
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"source/byte_io.h",
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"source/fec_receiver_impl.cc",
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"source/fec_receiver_impl.h",
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"source/receive_statistics_impl.cc",
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"source/receive_statistics_impl.h",
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"source/remote_ntp_time_estimator.cc",
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"source/rtp_header_parser.cc",
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"source/rtp_rtcp_config.h",
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"source/rtp_rtcp_impl.cc",
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"source/rtp_rtcp_impl.h",
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"source/rtcp_packet.cc",
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"source/rtcp_packet.h",
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"source/rtcp_receiver.cc",
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"source/rtcp_receiver.h",
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"source/rtcp_receiver_help.cc",
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"source/rtcp_receiver_help.h",
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"source/rtcp_sender.cc",
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"source/rtcp_sender.h",
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"source/rtcp_utility.cc",
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"source/rtcp_utility.h",
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"source/rtp_header_extension.cc",
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"source/rtp_header_extension.h",
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"source/rtp_receiver_impl.cc",
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"source/rtp_receiver_impl.h",
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"source/rtp_sender.cc",
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"source/rtp_sender.h",
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"source/rtp_utility.cc",
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"source/rtp_utility.h",
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"source/ssrc_database.cc",
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"source/ssrc_database.h",
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"source/tmmbr_help.cc",
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"source/tmmbr_help.h",
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# Audio Files
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"source/dtmf_queue.cc",
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"source/dtmf_queue.h",
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"source/rtp_receiver_audio.cc",
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"source/rtp_receiver_audio.h",
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"source/rtp_sender_audio.cc",
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"source/rtp_sender_audio.h",
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# Video Files
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"source/fec_private_tables_random.h",
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"source/fec_private_tables_bursty.h",
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"source/forward_error_correction.cc",
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"source/forward_error_correction.h",
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"source/forward_error_correction_internal.cc",
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"source/forward_error_correction_internal.h",
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"source/producer_fec.cc",
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"source/producer_fec.h",
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"source/rtp_packet_history.cc",
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"source/rtp_packet_history.h",
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"source/rtp_payload_registry.cc",
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"source/rtp_receiver_strategy.cc",
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"source/rtp_receiver_strategy.h",
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"source/rtp_receiver_video.cc",
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"source/rtp_receiver_video.h",
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"source/rtp_sender_video.cc",
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"source/rtp_sender_video.h",
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"source/video_codec_information.h",
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"source/rtp_format.cc",
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"source/rtp_format.h",
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"source/rtp_format_h264.cc",
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"source/rtp_format_h264.h",
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"source/rtp_format_vp8.cc",
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"source/rtp_format_vp8.h",
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"source/rtp_format_video_generic.cc",
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"source/rtp_format_video_generic.h",
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"source/vp8_partition_aggregator.cc",
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"source/vp8_partition_aggregator.h",
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# Mocks
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"mocks/mock_rtp_rtcp.h",
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"source/mock/mock_rtp_payload_strategy.h",
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]
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configs += [ "../..:common_config" ]
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public_configs = [ "../..:common_inherited_config" ]
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if (is_clang) {
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# Suppress warnings from Chrome's Clang plugins.
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# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
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configs -= [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../../system_wrappers",
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"../pacing",
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"../remote_bitrate_estimator",
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]
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if (is_win) {
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cflags = [
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# TODO(jschuh): Bug 1348: fix this warning.
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"/wd4267", # size_t to int truncations
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]
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}
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}
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