[LAD] Re: Direct Stream Digital / Pulse Density Modulation musing/questions

Maitland Vaughan-Turner maitlandvt at gmail.com
Tue Sep 25 04:41:39 UTC 2007

On 9/24/07, linux-audio-dev-request at lists.linuxaudio.org

> Date: Mon, 24 Sep 2007 21:59:01 +0200
> From: Fons Adriaensen <fons at kokkinizita.net>
> On Mon, Sep 24, 2007 at 11:50:55AM -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?

> > 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.

> 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.

Like, if I were just recording an acoustic guitar and vocals, of
course 24 bit would be the best choice.

But if I'm recording a live band, there is just so much stuff
happening at once...  You can't pinpoint an exact time when the
keyboard player presses the key, and you can't pinpoint just when I
pluck that bass string.  A 96 khz 24bit system might say that the two
events happened at exactly the same time, when really it was closer to
1/100000 of a second apart.  Now think about how many times something
like that could happen in a live recording with many instruments and
vocals and background noise from the crowd, etc.   I'd rather have the
detail than the dynamic range in that case...


More information about the Linux-audio-dev mailing list