63 lines
2.2 KiB
C++
Executable File
63 lines
2.2 KiB
C++
Executable File
/**********
|
|
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) 1996-2016 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";
|
|
}
|
|
|