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Guest
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 Posted: Tue Feb 06, 2007 12:28 am Post subject: Interesting technical question |
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Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of
44.1k samples/second, and you want to cut the size of the file in half. You
have 2 choices: 64kb/s at the same sample rate, or throwing away half the
samples, to 22.05k samples/second. Which choice would the average person
prefer? Assume normal hearing.
Thanks,
Norm Strong |
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Richard Crowley
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| Posted: Tue Feb 06, 2007 12:44 am Post subject: Re: Interesting technical question |
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<normanstrong@comcast.net> wrote ...
| Quote: | Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate
of 44.1k samples/second, and you want to cut the size of the file in half.
You have 2 choices: 64kb/s at the same sample rate, or throwing away half
the samples, to 22.05k samples/second. Which choice would the average
person prefer? Assume normal hearing.
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Do you think there is a single answer for all types of content
(music, speech, etc.) and for all situations? If there were,
why would we need separate control over all those parameters
(and even others)? |
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Todd H.
Guest
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| Posted: Tue Feb 06, 2007 1:31 am Post subject: Re: Interesting technical question |
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<normanstrong@comcast.net> writes:
| Quote: | Let's say you have an mp3 file at a bit rate of 128kb/s and a sample rate of
44.1k samples/second, and you want to cut the size of the file in half. You
have 2 choices: 64kb/s at the same sample rate, or throwing away half the
samples, to 22.05k samples/second.
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But it's not like that, actually. You appear to be applying linear
PCM thinking here to a concept that involves lossy compression.
Your question isn't terribly well constained so it's difficult to
answer.
One thing worth noting though, if you reduce the front end sample rate
to 22k, Nyquist demands that you have to say goodbye to anthing near
11kHz or above straight away by low pass filtering the source.
Listeners will have a big problem with that if we're talking about a
general musical source, for instance.
Also, dropping samples isn't a good way to do it. If you just
casually throw away half the samples yet your analog input wasn't low
pass filtered adequate to ensure the Nyquist criteria was met, you are
nearly guaranteed to be introducing new frequencies into your sampled
material through aliasing. Listeners hate that too.
There are lots of ways to get a file size in half. Choice of codec,
parameters provided to the codec, variable bit rate techniques,
encoding mono vs stereo... No one size fits all surely.
What problem are you trying to solve?
Best Regards,
--
/"\ ASCII Ribbon Campaign | Todd H
\ / | http://www.toddh.net/
X Promoting good netiquette | http://triplethreatband.com/
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