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--></style><title>Re: [Live-devel] Queries related to Streaming MP3
clips us</title></head><body>
<blockquote type="cite" cite><font face="Tahoma" size="-1">I
have some doubts regarding the RTP Payload for
MP3.</font></blockquote>
<blockquote type="cite" cite> </blockquote>
<blockquote type="cite" cite><font face="Tahoma" size="-1">1) When I
received the payload it's something like
this </font></blockquote>
<blockquote type="cite" cite><font face="Tahoma" size="-1">
00 00 00 00 ff e3 e8 .....</font></blockquote>
<blockquote type="cite" cite><font face="Tahoma"
size="-1"> The valid data starts only after 4 bytes
in all the cases. Why is that so?</font></blockquote>
<div><br></div>
<div>Because these first 4 bytes are the "MPEG Audio-specific
header" defined for the RTP payload format for MPEG-1 or 2
Elementary Stream audio (which includes MP3), defined in RFC 2250 (see
section 3.5).</div>
<div><br></div>
<div>If you transmit your data using the "MPEG1or2AudioRTPSink"
class, and receive it using the "MPEG1or2AudioRTPSource"
class, then the inserting and removing of this header is done
automatically, and you don't have to know or care about it.</div>
<div><br></div>
<div>See the code for the "testMP3Streamer" and
"testMP3Receiver" for examples of how this is done (over
multicast, in this case).</div>
<x-sigsep><pre>--
</pre></x-sigsep>
<div><br>
Ross Finlayson<br>
Live Networks, Inc.<br>
http://www.live555.com/</div>
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