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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.
// AC3 Audio RTP Sources
// Implementation
#include "AC3AudioRTPSource.hh"
AC3AudioRTPSource*
AC3AudioRTPSource::createNew(UsageEnvironment& env,
Groupsock* RTPgs,
unsigned char rtpPayloadFormat,
unsigned rtpTimestampFrequency) {
return new AC3AudioRTPSource(env, RTPgs, rtpPayloadFormat,
rtpTimestampFrequency);
}
AC3AudioRTPSource::AC3AudioRTPSource(UsageEnvironment& env,
Groupsock* rtpGS,
unsigned char rtpPayloadFormat,
unsigned rtpTimestampFrequency)
: MultiFramedRTPSource(env, rtpGS,
rtpPayloadFormat, rtpTimestampFrequency) {
}
AC3AudioRTPSource::~AC3AudioRTPSource() {
}
Boolean AC3AudioRTPSource
::processSpecialHeader(BufferedPacket* packet,
unsigned& resultSpecialHeaderSize) {
unsigned char* headerStart = packet->data();
unsigned packetSize = packet->dataSize();
// There's a 1-byte "NDU" header, containing the number of frames
// present in this RTP packet.
if (packetSize < 2) return False;
unsigned char numFrames = headerStart[0];
if (numFrames == 0) return False;
// TEMP: We can't currently handle packets containing > 1 frame #####
if (numFrames > 1) {
envir() << "AC3AudioRTPSource::processSpecialHeader(): packet contains "
<< numFrames << " frames (we can't handle this!)\n";
return False;
}
// We current can't handle packets that consist only of redundant data:
unsigned char typ_field = headerStart[1] >> 6;
if (typ_field >= 2) return False;
fCurrentPacketBeginsFrame = fCurrentPacketCompletesFrame;
// whether the *previous* packet ended a frame
// The RTP "M" (marker) bit indicates the last fragment of a frame:
fCurrentPacketCompletesFrame = packet->rtpMarkerBit();
resultSpecialHeaderSize = 2;
return True;
}
char const* AC3AudioRTPSource::MIMEtype() const {
return "audio/AC3";
}
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