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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.
// RTP sink for GSM audio
// Implementation
#include "GSMAudioRTPSink.hh"
GSMAudioRTPSink::GSMAudioRTPSink(UsageEnvironment& env, Groupsock* RTPgs)
: AudioRTPSink(env, RTPgs, 3, 8000, "GSM") {
}
GSMAudioRTPSink::~GSMAudioRTPSink() {
}
GSMAudioRTPSink*
GSMAudioRTPSink::createNew(UsageEnvironment& env, Groupsock* RTPgs) {
return new GSMAudioRTPSink(env, RTPgs);
}
Boolean GSMAudioRTPSink
::frameCanAppearAfterPacketStart(unsigned char const* /*frameStart*/,
unsigned /*numBytesInFrame*/) const {
// Allow at most 5 frames in a single packet:
return numFramesUsedSoFar() < 5;
}
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