Subject: Re: [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions
From: Anders Torger (torger_AT_ludd.luth.se)
Date: Mon Jul 28 2003 - 15:00:41 EEST
It becomes even more interesting when you bring in dither. Then you can
represent signals whose amplitude is less than one bit, and you can
increase time resolution to infinity.
Also, DSD is so cumbersome to process, that it is often converted to and
from PCM during production to be able to apply digital effects and
stuff like that. So it is very likely that an SACD you buy has been
through PCM somewhere in the production.
Audibility differences between DSD and PCM are more or less meaningless
to discuss, it is next to impossible to make a correct comparison, and
differences are so small that almost no-one is able to hear them. Most
cannot reliably hear a difference between 44.1 and 96 kHz. I know I
can't. Increasing the dynamic range from 16 to 24 bits is a much
greater win than increasing the sample rate.
Sony and others has put out straight lies to the public in the
commercials where they compare DSD and PCM, which is one reason why
there are so many myths about DSD vs PCM around.
/Anders Torger
On Monday 28 July 2003 10.58, Denis Sbragion wrote:
> Hello Michael,
>
> At 10.13 28/07/2003 +0200, you wrote:
> ...
>
> >A 22050 Hz sine could be really accurate (one sample up, one sample
> > down every 1/22050th second), and so is 11025. But intermediary
> > frequencies introduces temporal aliasing, some metallic feeling due
> > to temporal quantization. This is inherent to the very low sampling
> > rate (96 kHz is just
>
> ...
>
>
> sorry, but this is incorrect. The sampled signal has nothing to do
> with the information it carries. The sampled signal just contains the
> information needed to get back the original signal but it isn't the
> original signal (despite at lower frequency it pretty much resemble
> the original signal), you have to pass it through a reconstruction
> filter to get the original (bandlimited) signal back. You can't
> simply look at the sampled signal to see what the original signal
> was, much the same way you can't just look at a compressed file to
> see what the original file was. This is one of the most common
> misconception circulating about PCM, but it is indeed a
> misconception.
> If you hear a difference between DSD and PCM the problem
> should be somewhere else, may be in the ADC/DAC used, may be in the
> DSD itself, but sure not in what you describe above.
>
> Bye,
> --
> Denis Sbragion
> InfoTecna
> Tel: +39 0362 805396, Fax: +39 0362 805404
> URL: http://www.infotecna.it
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