[LAD] [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

Maitland Vaughan-Turner maitlandvt at gmail.com
Mon Sep 24 01:06:07 UTC 2007

> On Tue, 22 Apr 2003 19:54:09 -0500
> "Dustin Barlow" <duslow at hotmail.com> wrote:
> > I read an interesting article on Direct Stream Digital (DSD) / Pulse Density
> > Modulation (PDM) entitled "A Better Mousetrap" by Brian Smithers in the May
> > 2003 issue of Electronic Musician.  Since, Brian did a good job explaining
> > PDM/DSD in quasi-layman terms, I'll just quote snippets from his article to
> > set the stage for my questions.
> <snip>
> > DSD/PDM appears to be a superiour technique for recording and playing audio
> > material.
> Having been around digital audio and digital signal processing for over 10
> years, I am still far from convinced.
> > Granted, this technology may never catch on because of all the
> > hardware and software changes that would be required to mirror what a
> > typical PCM based DAW currently does.  But, if DSD/PDM does catch on, and
> > DAWs start being produced, how will this effect current audio DSP
> > techniques?
> I have not looked into the maths behind algorithm development in DSD/PDM,
> but I doubt it is anywhere near as easy as with PCM.
> > The article mentions a program called Pyramix (Windows) which features DSD
> > support.  However, for Pyramix to do EQ, dynamics, reverb processing, and to
> > display waveforms and vu levels, it converts DSD to a "high quality" PCM
> > format.
> That should tell you something :-).


So..?  Most PCM converters utilize a 1-bit stream also.  Why not
utilize all the tools available for the task at hand?

As for processing, you can look at a PCM representation of a waveform
to ease the processing load and then just apply the changes to the
orignal DSD stream without ever having to process in the 1-bit domain
directly (which is way more processor intensive since you have to look
at a huge chunk of the stream in order to extract the amplitude data
that is available in each multi-bit sample).

IMHO, though, the hippest alternative at present is to process a DSD
stream in the analog domain and re-record it to DSD.  This results in
a very "analog" sound.  These days you can get analog gear with a
respectable dynamic range for a song (Mackie Onyx anyone?).  When you
can get a 130 dB S/N ratio in the analog domain you really don't lose
too much converting back and forth from 1-bit domain.  It's freakin

 If you haven't tried recording 1-bit.  Do yourself a favor and demo
one of the new Korg recorders.  It really is really good, no kidding.


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