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📄 mpeg1or2audiortpsink.hh

📁 流媒体传输协议的实现代码,非常有用.可以支持rtsp mms等流媒体传输协议
💻 HH
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/**********This library is free software; you can redistribute it and/or modify it underthe terms of the GNU Lesser General Public License as published by theFree Software Foundation; either version 2.1 of the License, or (at youroption) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)This library is distributed in the hope that it will be useful, but WITHOUTANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESSFOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License formore details.You should have received a copy of the GNU Lesser General Public Licensealong with this library; if not, write to the Free Software Foundation, Inc.,59 Temple Place, Suite 330, Boston, MA  02111-1307  USA**********/// "liveMedia"// Copyright (c) 1996-2004 Live Networks, Inc.  All rights reserved.// RTP sink for MPEG audio (RFC 2250)// C++ header#ifndef _MPEG_1OR2_AUDIO_RTP_SINK_HH#define _MPEG_1OR2_AUDIO_RTP_SINK_HH#ifndef _AUDIO_RTP_SINK_HH#include "AudioRTPSink.hh"#endifclass MPEG1or2AudioRTPSink: public AudioRTPSink {public:  static MPEG1or2AudioRTPSink* createNew(UsageEnvironment& env,				     Groupsock* RTPgs);protected:  MPEG1or2AudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs);	// called only by createNew()  virtual ~MPEG1or2AudioRTPSink();private: // redefined virtual functions:  virtual void doSpecialFrameHandling(unsigned fragmentationOffset,                                      unsigned char* frameStart,                                      unsigned numBytesInFrame,                                      struct timeval frameTimestamp,                                      unsigned numRemainingBytes);  virtual unsigned specialHeaderSize() const;};#endif

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