07bf43c673
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches. In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on. BUG= TEST=vie/voe_auto_test, trybots Review URL: https://webrtc-codereview.appspot.com/1001006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
101 lines
2.9 KiB
Python
101 lines
2.9 KiB
Python
# Copyright (c) 2011 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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{
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'targets': [
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{
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'target_name': 'rtp_rtcp',
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'type': '<(library)',
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'dependencies': [
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
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'<(webrtc_root)/modules/modules.gyp:paced_sender',
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],
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'include_dirs': [
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'../interface',
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'../../interface',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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'../interface',
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'../../interface',
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],
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},
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'sources': [
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# Common
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'../interface/rtp_rtcp.h',
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'../interface/rtp_rtcp_defines.h',
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'bitrate.cc',
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'Bitrate.h',
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'rtp_rtcp_config.h',
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'rtp_rtcp_impl.cc',
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'rtp_rtcp_impl.h',
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'rtcp_receiver.cc',
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'rtcp_receiver.h',
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'rtcp_receiver_help.cc',
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'rtcp_receiver_help.h',
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'rtcp_sender.cc',
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'rtcp_sender.h',
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'rtcp_utility.cc',
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'rtcp_utility.h',
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'rtp_header_extension.cc',
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'rtp_header_extension.h',
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'rtp_receiver.cc',
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'rtp_receiver.h',
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'rtp_sender.cc',
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'rtp_sender.h',
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'rtp_utility.cc',
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'rtp_utility.h',
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'ssrc_database.cc',
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'ssrc_database.h',
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'tmmbr_help.cc',
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'tmmbr_help.h',
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# Audio Files
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'dtmf_queue.cc',
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'dtmf_queue.h',
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'rtp_receiver_audio.cc',
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'rtp_receiver_audio.h',
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'rtp_sender_audio.cc',
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'rtp_sender_audio.h',
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# Video Files
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'fec_private_tables_random.h',
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'fec_private_tables_bursty.h',
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'forward_error_correction.cc',
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'forward_error_correction.h',
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'forward_error_correction_internal.cc',
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'forward_error_correction_internal.h',
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'producer_fec.cc',
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'producer_fec.h',
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'rtp_packet_history.cc',
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'rtp_packet_history.h',
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'rtp_receiver_strategy.cc',
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'rtp_receiver_stragegy.h',
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'rtp_receiver_video.cc',
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'rtp_receiver_video.h',
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'rtp_sender_video.cc',
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'rtp_sender_video.h',
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'receiver_fec.cc',
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'receiver_fec.h',
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'video_codec_information.h',
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'rtp_format_vp8.cc',
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'rtp_format_vp8.h',
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'vp8_partition_aggregator.cc',
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'vp8_partition_aggregator.h',
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# Mocks
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'../mocks/mock_rtp_rtcp.h',
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], # source
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},
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],
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}
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# Local Variables:
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# tab-width:2
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# indent-tabs-mode:nil
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# End:
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# vim: set expandtab tabstop=2 shiftwidth=2:
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