blob: faa1175d8722ce4731ae2926f4959ec172f98726 [file] [log] [blame]
/**********
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
Free Software Foundation; either version 3 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
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
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
**********/
// "liveMedia"
// Copyright (c) 1996-2020 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