1 | /********** |
2 | This library is free software; you can redistribute it and/or modify it under |
3 | the terms of the GNU Lesser General Public License as published by the |
4 | Free Software Foundation; either version 3 of the License, or (at your |
5 | option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.) |
6 | |
7 | This library is distributed in the hope that it will be useful, but WITHOUT |
8 | ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS |
9 | FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for |
10 | more details. |
11 | |
12 | You should have received a copy of the GNU Lesser General Public License |
13 | along with this library; if not, write to the Free Software Foundation, Inc., |
14 | 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
15 | **********/ |
16 | // "liveMedia" |
17 | // Copyright (c) 1996-2020 Live Networks, Inc. All rights reserved. |
18 | // MPEG-1 or MPEG-2 Audio RTP Sources |
19 | // Implementation |
20 | |
21 | #include "MPEG1or2AudioRTPSource.hh" |
22 | |
23 | MPEG1or2AudioRTPSource* |
24 | MPEG1or2AudioRTPSource::createNew(UsageEnvironment& env, |
25 | Groupsock* RTPgs, |
26 | unsigned char rtpPayloadFormat, |
27 | unsigned rtpTimestampFrequency) { |
28 | return new MPEG1or2AudioRTPSource(env, RTPgs, rtpPayloadFormat, |
29 | rtpTimestampFrequency); |
30 | } |
31 | |
32 | MPEG1or2AudioRTPSource::MPEG1or2AudioRTPSource(UsageEnvironment& env, |
33 | Groupsock* rtpGS, |
34 | unsigned char rtpPayloadFormat, |
35 | unsigned rtpTimestampFrequency) |
36 | : MultiFramedRTPSource(env, rtpGS, |
37 | rtpPayloadFormat, rtpTimestampFrequency) { |
38 | } |
39 | |
40 | MPEG1or2AudioRTPSource::~MPEG1or2AudioRTPSource() { |
41 | } |
42 | |
43 | Boolean MPEG1or2AudioRTPSource |
44 | ::(BufferedPacket* packet, |
45 | unsigned& ) { |
46 | // There's a 4-byte header indicating fragmentation. |
47 | if (packet->dataSize() < 4) return False; |
48 | |
49 | // Note: This fragmentation header is actually useless to us, because |
50 | // it doesn't tell us whether or not this RTP packet *ends* a |
51 | // fragmented frame. Thus, we can't use it to properly set |
52 | // "fCurrentPacketCompletesFrame". Instead, we assume that even |
53 | // a partial audio frame will be usable to clients. |
54 | |
55 | resultSpecialHeaderSize = 4; |
56 | return True; |
57 | } |
58 | |
59 | char const* MPEG1or2AudioRTPSource::MIMEtype() const { |
60 | return "audio/MPEG" ; |
61 | } |
62 | |
63 | |