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GSMAudioRTPSink.hh
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1/**********
2This library is free software; you can redistribute it and/or modify it under
3the terms of the GNU Lesser General Public License as published by the
4Free Software Foundation; either version 3 of the License, or (at your
5option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
6
7This library is distributed in the hope that it will be useful, but WITHOUT
8ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
9FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
10more details.
11
12You should have received a copy of the GNU Lesser General Public License
13along with this library; if not, write to the Free Software Foundation, Inc.,
1451 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
15**********/
16// "liveMedia"
17// Copyright (c) 1996-2024 Live Networks, Inc. All rights reserved.
18// RTP sink for GSM audio
19// C++ header
20
21#ifndef _GSM_AUDIO_RTP_SINK_HH
22#define _GSM_AUDIO_RTP_SINK_HH
23
24#ifndef _AUDIO_RTP_SINK_HH
25#include "AudioRTPSink.hh"
26#endif
27
29public:
31
32protected:
34 // called only by createNew()
35
37
38private: // redefined virtual functions:
39 virtual
40 Boolean frameCanAppearAfterPacketStart(unsigned char const* frameStart,
41 unsigned numBytesInFrame) const;
42};
43
44#endif
unsigned char Boolean
Definition: Boolean.hh:25
static GSMAudioRTPSink * createNew(UsageEnvironment &env, Groupsock *RTPgs)
GSMAudioRTPSink(UsageEnvironment &env, Groupsock *RTPgs)
virtual Boolean frameCanAppearAfterPacketStart(unsigned char const *frameStart, unsigned numBytesInFrame) const
virtual ~GSMAudioRTPSink()