[wplug] MP3 or OGG-Vorbis recovery tool?

[bgtrio at yahoo.com]

You definitely can not get the original waveform back once you've encoded
in mp3, just something that sounds like it to the human ear.  The
compression works by discarding (mostly) inaudible.  I've heard that
choral music doesn't compress as well as most rock music because the
layers of similar sounds are more important there, and a lot fo that
information is lost.  It's kind of like a movie facade- if you just look
at it from the front it looks real but once you get behind it you can see
that there's nothing inside it.  The trick is to make sure you only see it
from the front.

So long story short, the waveform you get from the mp3 or ogg compressed
file is quite different from the original, it just fools your ears.

If you want lossless compression try flac.

Bryguy

http://www.livejournal.com/~bryguypgh

On Mon, 27 Oct 2003, K G wrote:

> On Mon, Oct 27, 2003 at 02:26:22PM -0500, Bill Moran wrote:
> > Not really.  .wav files are made by sampling the waveform at fixed
> > intervals (44khz, right?), whereas mp3s use more complex mathematics
> > to store the data.  I don't have a great understanding of how the mp3
> > math works, but I've been lead to believe that it's similar in concept
> > to how jpeg files store images.  If that is correct, here is what your
> > circumstance is:
> >
> > You start with a .wav (whether it's straigh from a CD, or in a .wav file,
> > it's in the same format) and you convert it to mp3 by changing the data
> > format and discarding some details that you'd never hear anyway, this
> > results in excellent compression.  Now you convert the mp3 back to .wav
> > format.  The .wav is exactly the same waveform representation as the
> > mp3, but is not the same as the original .wav.
>
> What I understand is that you do a Fourier transform (i.e. you convert
> the data to something more like (x frequency, y amplitude) datapoints.
> This results in incredibly good sound, but very little compression. So
> then what you do is discard some of the datapoints (like if you have a
> high amplitude wave and a low amplitude at a similar frequency, people
> are probably not going to hear the quieter wave). Anyway, that's where
> you loose data. Then this is compressed using a lossless compression
> algorithm.
>
> To answer the original question, I don't know. If you reencode the wave
> files, the question becomes "will the algorithm drop more data?". I think
> that the answer is using the same algorithm, it shouldn't drop much (but
> might drop some). One of my professors did an experiment with a few
> photographs and jpeg and the result was that there were a few extra
> pixels wrong on the second encoding (I don't remember the resolution,
> but it was more than 1024x1024).
>
> Kuzman
>