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GSMAudioRTPSink.hh
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1 /**********
2 This library is free software; you can redistribute it and/or modify it under
3 the terms of the GNU Lesser General Public License as published by the
4 Free Software Foundation; either version 3 of the License, or (at your
5 option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)
6 
7 This library is distributed in the hope that it will be useful, but WITHOUT
8 ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
9 FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for
10 more details.
11 
12 You should have received a copy of the GNU Lesser General Public License
13 along with this library; if not, write to the Free Software Foundation, Inc.,
14 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
15 **********/
16 // "liveMedia"
17 // Copyright (c) 1996-2021 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 
29 public:
31 
32 protected:
34  // called only by createNew()
35 
36  virtual ~GSMAudioRTPSink();
37 
38 private: // 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
GSMAudioRTPSink(UsageEnvironment &env, Groupsock *RTPgs)
virtual Boolean frameCanAppearAfterPacketStart(unsigned char const *frameStart, unsigned numBytesInFrame) const
virtual ~GSMAudioRTPSink()
static GSMAudioRTPSink * createNew(UsageEnvironment &env, Groupsock *RTPgs)