🚀 go-pugleaf

zawada@EN.ECN.PURDUE.EDU (Paul J Zawada) writes:


>A day or two ago, I heard another one bit D-to-A algorithm from a
>strictly unreliable source.  (A musician friend of mine.)  Since
>1 bit D-to-A was a recent topic, I thought I'd share this one with 
>with the net.  He claims this new revolutionary algorithm uses one
>bit to tell whether the output voltage should be incremented or
>decremented from its current state.

Sounds like a delta modulator (not new).

>Now I'm only an undergrad in EE and I haven't studied dsp much, but
>I cannot see how to actually implement this.  I see a few problems.

Input voltage is tracked on a capacitor.  Capacitor voltage is compared
to input.  Comparator output dumps fixed current on/off the capacitor.

-- 

	Jeff Mulligan (jbm@aurora.arc.nasa.gov)
	NASA/Ames Research Ctr., Mail Stop 239-3, Moffet Field CA, 94035
	(415) 694-3745

In article <8911201702.AA03844@en.ecn.purdue.edu> zawada@EN.ECN.PURDUE.EDU (Paul J Zawada) writes:
>
>A day or two ago, I heard another one bit D-to-A algorithm from a
>strictly unreliable source.  (A musician friend of mine.)  Since
>1 bit D-to-A was a recent topic, I thought I'd share this one with 
>with the net.  He claims this new revolutionary algorithm uses one
>bit to tell whether the output voltage should be incremented or
>decremented from its current state.  i.e. Voltages are not quantisized

But this is good old delta modulation, one of the very first digital modulation
systems invented.  

It works quite well provided you choose a bit rate that is high enough to 
handle the highest expected slope in the input signal.

By the way I understand that NASA used it back in the Apollo days as the last
ditch communications system - the last few minutes before complete comms
blackout during re-entry.


Alistair URIE                                     Radio and Satellite Networks
Phone:   +61 3 541 6370                           Telecom Research Laboratories
Fax:     +61 3 543 3339                           770 Blackburn Rd. Clayton Vic
Internet: aurie@rhea.trl.oz.au                                        AUSTRALIA

Yes this sounds like delta modulation but that is not new. The only recent
anouncement that concerns a 1 bit D/A is from Phillips addressing the
problem of costly and inaccurate 16/18 + D/A's on low level signals
in CD players. They claim that by drastically oversampling at x256 they
convert the parallel words to an 11.3MHz bit stream that represents a
"pulse density modulation" that is similar to delta modulation in that
it pumps up and down a filter. The rf components can be removed by gentle
filtering because there is a wide frequency difference between the information
and the data rate. Also adverse effects on the phase of inband signals are 
minimized. Since only two states are needed to go up or down a simple 1 bit
D/A can be used and the filter designed with switched capacitor technology.
Using VLSI, two chips can do the servo control, digital detection, correction,
interpolation, oversampling, and D/A conversion for a fraction of the cost
of todays players while maintaining the same S/N and better low level
linearity.

Isn't a 1 bit D/A just a delta modulator? There have a few products
based on delta mod in the pro music field, most notably the TC2900 delay
line. It samples at 100Mhz (so they claim). It DOES sound wonderful.

-- 
_______________________________________________________________________________
Chris Muir                             |   "There is no language in our lungs
{hplabs,pacbell,ucbvax,apple}          |    to tell the world just how we feel" 
!well!cbm                              |                         - A. Partridge

This general family of conversions is known as Delta Modulation. As Paul Zawada
says, you use a single bit to indicate whether the output should be incremented
or decremented.  The nice thing about the method is the simplicity of the
conversions:

	On A->D, if the analog signal is greater (less than) the reference, 
	then output a 1 (0) and increment (decrement) the value of the 
	reference before the next sampling.

	On D->A, if the input is 1 (0), then increment (decrement) the 
	counter and send the result to the DAC.

If the signal has not changed at all, the result is least significant bit
jitter as the reference signal goes just above, then just below the actual
input.

As you can see, this method eliminates successive aproximation loops and other
complications. For the method to be successful, it is essential that the 
change in the signal be very small between samples. Normally this means
that the sampling rate is much higher than with SA or flash conversion
techniques.

An ordinary converter at 44 KHz could change from one extreme value to the
other from one sample to the next.  This occurs only on the edge of Nyquists
limit, and the presence of such changes makes it likely that signals over the
limit could occur. Usually, signals change in an orderly manner.  If that
converter wasa 16 bit SA converter, the A->D was actually operating at about
 .7Mhz.  A DM converter at .7Mhz could, in the same interval, only change by
 16 samples. For a full-scale triangle wave input, the maximum frequency which
 can be followed is 10 Hz. 

Usually, the situation is much better, but this inefficiency is one of the
reasons for the developement of Adaptive Delta Modulation techniqes.
Essentially, these are two-bit converters: one bit says whether the value
should be increased or decreased, and the other says whether the size of the
increment should be increased or decreased.

Thus while the basic DM degenerates to a ramp, the ADM degenerates to something
closer to successive approximation.  With these variations, the joints benefits
of (relatively) simple one stage converters and decreased data storage 
requirements are obtained.  If you want to process the music, as in a
synthesizer or effects box, you're better off with parallel techniques. If your
application calls for the storage of data in compressed form, and especially
if your medium is serial to start with ADM type methods are ideal.  That's
why it was used in (I think it was the ..) DBX 700, an Industrial quality
digital audio system which stored stereo audio of very high quality on a
consumer Beta or VHS video tape, disguised as a video signal.