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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
- 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) 1996-2019 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
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