GSMAudioRTPSink.hh

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/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
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This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.
 
You should have received a copy of the GNU Lesser General Public License
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**********/
// "liveMedia"
// Copyright (c) 1996-2026 Live Networks, Inc.  All rights reserved.
// RTP sink for GSM audio
// C++ header
 
#ifndef _GSM_AUDIO_RTP_SINK_HH
#define _GSM_AUDIO_RTP_SINK_HH
 
#ifndef _AUDIO_RTP_SINK_HH
#include "AudioRTPSink.hh"
#endif
 
class GSMAudioRTPSink: public AudioRTPSink {
public:
  static GSMAudioRTPSink* createNew(UsageEnvironment& env, Groupsock* RTPgs);
 
protected:
  GSMAudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs);
        // called only by createNew()
 
  virtual ~GSMAudioRTPSink();
 
private: // redefined virtual functions:
  virtual
  Boolean frameCanAppearAfterPacketStart(unsigned char const* frameStart,
                                         unsigned numBytesInFrame) const;
};
 
#endif