/usr/include/liveMedia/VorbisAudioRTPSink.hh is in liblivemedia-dev 2014.01.13-1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
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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 2.1 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) 1996-2014 Live Networks, Inc. All rights reserved.
// RTP sink for Vorbis audio
// C++ header
#ifndef _VORBIS_AUDIO_RTP_SINK_HH
#define _VORBIS_AUDIO_RTP_SINK_HH
#ifndef _AUDIO_RTP_SINK_HH
#include "AudioRTPSink.hh"
#endif
class VorbisAudioRTPSink: public AudioRTPSink {
public:
static VorbisAudioRTPSink* createNew(UsageEnvironment& env,
Groupsock* RTPgs,
u_int8_t rtpPayloadFormat, u_int32_t rtpTimestampFrequency, unsigned numChannels,
// The following headers provide the 'configuration' information, for the SDP description:
u_int8_t* identificationHeader, unsigned identificationHeaderSize,
u_int8_t* commentHeader, unsigned commentHeaderSize,
u_int8_t* setupHeader, unsigned setupHeaderSize);
static VorbisAudioRTPSink* createNew(UsageEnvironment& env,
Groupsock* RTPgs,
u_int8_t rtpPayloadFormat, u_int32_t rtpTimestampFrequency, unsigned numChannels,
char const* configStr);
// an optional variant of "createNew()" that takes a Base-64-encoded 'configuration' string,
// rather than the raw configuration headers. as parameter.
protected:
VorbisAudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs,
u_int8_t rtpPayloadFormat, u_int32_t rtpTimestampFrequency, unsigned numChannels,
u_int8_t* identificationHeader, unsigned identificationHeaderSize,
u_int8_t* commentHeader, unsigned commentHeaderSize,
u_int8_t* setupHeader, unsigned setupHeaderSize,
u_int32_t identField = 0xFACADE);
// called only by createNew()
virtual ~VorbisAudioRTPSink();
private: // redefined virtual functions:
virtual char const* auxSDPLine(); // for the "a=fmtp:" SDP line
virtual void doSpecialFrameHandling(unsigned fragmentationOffset,
unsigned char* frameStart,
unsigned numBytesInFrame,
struct timeval framePresentationTime,
unsigned numRemainingBytes);
virtual Boolean frameCanAppearAfterPacketStart(unsigned char const* frameStart,
unsigned numBytesInFrame) const;
virtual unsigned specialHeaderSize() const;
virtual unsigned frameSpecificHeaderSize() const;
#endif
private:
u_int32_t fIdent; // "Ident" field used by this stream. (Only the low 24 bits of this are used.)
char* fFmtpSDPLine;
};
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