On Mon, Sep 24, 2007 at 09:41:39PM -0700, Maitland Vaughan-Turner wrote:
> > > Intuitively, one could also say that more sample points yield a
> > > waveform that is closer to a continuous, analog waveform. Thus it
> > > sounds more analog.
> >
> > This is completely wrong. Sorry to be rude, but such a statement
> > only shows your lack of understanding.
>
> Why is it wrong? If I drew some dots on a waveform and then connected
> the dots, to try to reconstruct the waveform, wouldn't I get a better
> result with more dots?
If you just connect the dots, yes. But that's not how an analog waveform
is reconstructed from PCM samples. 'Connecting the dots' is not even part
of that process.
> > > Thanks for the link. My whole point of digging up this old thread
> > > though, was to say that I've tried it, and my ears tell me that the
> > > papers are incorrect.
> >
> > Then please point out the errors in the paper by Lipshitz and Vanderkooy.
>
> my ears tell me that... that's all; it's just subjective. haha, I see
> subjective reports don't get you far around here.
You said the papers were incorrect. Then point out the errors.
And indeed, a subjective evaluation is useless if not the result
of a double blind test.
If you have ever been involved in organising a controlled listening
test you should know how easy it is to get completely invalid results
and to fool yourself into believing things that are just an illusion.
> > I'm not saying that DSD is crap. It sounds well. But it doesn't meet
> > the claims set for it (as shown by L&V - you need at least two bits
> > to have a 'linear' channel) and as a storage or transmission format
> > it's inefficient compared to PCM. That means that if you use PCM with
> > the same number of bits per second as used by DSD, you get a better
> > result than what DSD delivers.
>
> well, what do you mean by better? It seems like 24 bit is already
> better in terms of dynamic range at any sample rate, but if you mean
> more detailed representation of a waveform (in time), it seems like
> you necessarily need to have the highest possible sample rate.
There is a point where more detail becomes irrelevant because it's
way below the noise. For a properly dithered PCM signal it can be
shown that the error that remains is in a strict mathematical sense
indistinguishable from noise. Hence if it's below the analog noise
floor it doeasn't matter any more. BUt you need at least _two_ bits
for this to work.
For 'detail in time' the situation is even simpler. If the waveform
is bandlimited, then *every* detail is captured by sampling it at a
rate equal to twice the bandwidth. Each sample does not only represent
the waveform at the time it was taken, but in fact contains information
about the entire waveform. The samples completely describe the waveform
in the same strict sense as three points are sufficient to define a
circle. It's not an approximation.
-- FA Follie! Follie! Delirio vano è questo ! _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev@email-addr-hidden http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-devReceived on Tue Sep 25 12:15:02 2007
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