/********** 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 2.1 of the License, or (at your option) any later version. (See .) 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) 1996-2015 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