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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.,
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
**********/
// "liveMedia"
// Copyright (c) 1996-2005 Live Networks, Inc. All rights reserved.
// MPEG-1 or MPEG-2 Audio RTP Sources
// Implementation
#include "MPEG1or2AudioRTPSource.hh"
MPEG1or2AudioRTPSource*
MPEG1or2AudioRTPSource::createNew(UsageEnvironment& env,
Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
unsigned rtpTimestampFrequency) {
return new MPEG1or2AudioRTPSource(env, RTPgs, rtpPayloadFormat,
rtpTimestampFrequency);
}
MPEG1or2AudioRTPSource::MPEG1or2AudioRTPSource(UsageEnvironment& env,
Groupsock* rtpGS,
unsigned char rtpPayloadFormat,
unsigned rtpTimestampFrequency)
: MultiFramedRTPSource(env, rtpGS,
rtpPayloadFormat, rtpTimestampFrequency) {
}
MPEG1or2AudioRTPSource::~MPEG1or2AudioRTPSource() {
}
Boolean MPEG1or2AudioRTPSource
::processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize) {
// There's a 4-byte header indicating fragmentation.
if (packet->dataSize() < 4) return False;
// Note: This fragmentation header is actually useless to us, because
// it doesn't tell us whether or not this RTP packet *ends* a
// fragmented frame. Thus, we can't use it to properly set
// "fCurrentPacketCompletesFrame". Instead, we assume that even
// a partial audio frame will be usable to clients.
resultSpecialHeaderSize = 4;
return True;
}
char const* MPEG1or2AudioRTPSource::MIMEtype() const {
return "audio/MPEG";
}
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