| /********** |
| 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 3 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) 1996-2020 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"; |
| } |
| |