[Imc-newsroom] mp3 upload problem solved (?)
Paul Riismandel
p-riism at uiuc.edu
Mon May 7 01:00:36 CDT 2001
I think I figured out why some MP3s upload fine, while others don't.
It all has to do with the size and bitrate of the file. I re-encoded the
FTAA Documentary mp3s at about 1/10 the data rate and file size and they
uploaded and work just fine. I don't know what the upper threshold is for
mp3 file size, but there are some basic guidelines you can follow that
should make things work.
When you're encoding .wav files into .mp3 you want to specify a relatively
low bitrate. There's two reasons for this, the first is so the server
won't balk, the second is so that the files are small enough for modem
users to download in a reasonable amount of time. When I looked at the
FTAA Documentary mp3 files on Sergei they were 320kbps (kbps = kilobits per
second)--more than 6 times the bandwidth of the fastest POTS modem! While
they sound great, they're too damn big (averaging 6 MB). Over a 56kbps
modem, expect this to take an hour or more (or at least 6x the real-time
length of the program).
I re-encoded the files at 24kbps and now they average about 600k
each. Over a decent 56kbps connection these can even be played in
real-time, without the user having to download the whole file before
listening. I also converted the files from stereo to mono before encoding,
because at low bitrates stereo sounds like shit.
If you're on Sergei and using MusicMatch Jukebox to encode your mp3s, you
can change your bitrates in the window where you select the files to
encode. On the bottom right of this window there's a little slider with a
number to the right. Move the slider to get the bitrate you want. It
seems to default to 96kbps, which is too big, so you'll always need to
change it.
Now, at low bitrates you do sacrifice overall fidelity and audio
quality. 16-24kbps mp3s sound like AM radio, but are quite intelligible
for speech, which is primarily what our stuff is. In mono 64 kbps is quite
high-fidelity, but takes 3-4 times longer to download and listen. In this
case we're trading fidelity for accessibility.
So here are the guidelines for encoding and upload mp3s to the IMC website:
To make your files listenable for almost any modem user (as low as 28.8 kpbs):
* Make sure your file is mono. If you're editing in Sound Forge and you
have a stereo file, select Process, then Channel Converter, then in the
Channel Converter window, select 'Stereo to Mono - Use both channels (50%)'
from the Name: pull-down menu.
* Encode at a bitrate of 16kbps -- no higher.
To make your files listenable for users on a 56kbps modem or better:
* Make sure your file is mono. If you're editing in Sound Forge and you
have a stereo file, select Process, then Channel Converter, then in the
Channel Converter window, select 'Stereo to Mono - Use both channels (50%)'
from the Name: pull-down menu.
* Encode at a bitrate of 16 to 20kbps -- no higher.
To make files that are closer to CD or 'broadcast quality' that radio
stations can download and use:
* Make sure your file is mono. If you're editing in Sound Forge and you
have a stereo file, select Process, then Channel Converter, then in the
Channel Converter window, select 'Stereo to Mono - Use both channels (50%)'
from the Name: pull-down menu.
* Encode at a bitrate of 48 to 64 kbps. 48kbps is the minimum level for
decent quality, making it a reasonable download for users on modem
connections (probably will take 2x real-time). 64 kbps is near-CD quality,
but increases your file size (and download time) by 33%.
And now, a quick FAQ:
Q: Why encode in mono?
A: As you might expect, decent quality stereo requires about twice as much
data than mono. At 24 kbps you essentially have two 12kbps mono tracks,
and 12kbps is too low to sound good at all. The basic lower threshold for
listenable quality in mp3 is 16kbps in mono, or 32kbps in stereo (which is
difficult or impossible to listen to in real-time on a modem connection).
Q: Why are we sacrificing audio quality for low bitrates?
A: High-quality takes long download times, and we cannot expect that our
viewers/listeners are on broadband connections like we have at the IMC (or
at the University). We are just limiting our audience if we ask a large
portion of our listeners to spend hours to download a half-hour piece that
we might otherwise make available for listening in real-time.
Q: What if a radio station wants to air our pieces from the website? Do we
want to force them to use AM radio-quality files?
A: If we expect that a file may be popular, I'd try uploading two
versions--one at low bitrate for modems, and one at high bitrates for
broadcast. I'd name the file with the bitrate appended to it (like:
ftaa_documentary_16kbps.mp3), and make sure that you mention the quality
level and bitrate in the summary when you publish.
I hope this is helpful, please post any questions. Though I'd prefer not
to argue the merits of mp3 quality, stereo vs. mono, etc., since what I've
posted above are fairly accepted standards, unless there's a magic bullets
that can give CD-quality at modem-accessible bitrates that I don't know about.
--Paul
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