[sdiy] Prophet VS a phase accumulator design

[karl dalen]

>Scott Nordlund <gsn10 at hotmail.com>:
> I think almost every sane (i.e. not Fairlight/Synclavier)
> polyphonic implementation is fixed rate, just some

> But that's almost saying that the only "true" variable 
> clock is a VCO. 

Thank you!

>Where I'm less clear is the Emulator/Emax stuff. Older
>variable rate samplers like the Fairlight required a
>separate sample RAM per voice, but the Emulator
>apparently kept the fixed sample rate.

Well, again i took the liberty this time to ask Dave Rossum
directly since you brought it up, i asked only about the Emus
but aparently he enjoyed the question so he just continued,
and thats great/splendid because he extended our discussion
by highlighting the next suspect, FZ1.. :)

----------------------------------------------------------------------
*The Emulator 1 used a high frequency varactor tuned (for pitch bend and
*vibrato) 10HMz oscillator, with programmable dividers to set the sample
*rate for each channel.  So they were variable sample rate channels,
*which produces no pitch-shifting aliasing.We also had a tracking lowpass
*filter, which reduced some of the imaging at low sample rates. The
*Emulator II worked in the same manner, although using a superior data
*encoding.  The original Emulator III operated similarly as well, using
*16 bit linear encoding, and a custom FIR digital oversampling filter
*chip ("F-chip") for each DAC to minimize imaging.

*The EMAX I, using our first VLSI chip - the "E-chip", was a hybrid
*approach.  It had an internal 10MHz clock, and 17 "logical" channels
*(one channel performed refresh and system access to the sound memory),
*which gave a channel service rate of about 588 KHz syncronous sample
*rate.  Each DAC was then operated at a jittered (drop sample) divisor
*of this rate.  If this seems hard to follow, it is.  IT was a very
*unusual approach that gave much lower pitch shift aliasing than most
*synchronous approaches of the time, yet had the advantage that
*synchronous systems of multiplexing the channel logic, though
*the output was not at a fixed sample rate.

*Beginning with the EMAX II and the Emulator IIIx, we used our "G-chip"
*technology, which was a 7th or 8th order interpolator operating from
*large fractional value phase accumulators.  These gave excellent (still
*largely unsurpassed) alias and image attenuation, with all the advantages
*of a syncronous design.  The original G-chip was 32 channels; the G-chip
*II had 64 channels per chip, and two chips could share a memory to give
*128 channels.

*I believe the Fairlight used a separate 6805 type processor, and a
*separate memory for each channel, and used a variable sample rate
*divisor output similar to the Emulator I (the Emulator I, though,
*shared the memory among all 8 channels, which gave it a huge cost
*advantage over the fairlight).  I don't know if the fairlight could
*do fine resolution vibrato or pitch bend, as this would require a
*VCO for the master oscillator.

*The original PPG was a synchronous drop-sample device, having all the
*aliasing problems this produces.So was the SP1200 and the Ensoniq.

*There was a 16 bit Casio machine (the first 16 bit sampler, I think)
*that used 2x oversampled linear interpolation.  I was impressed with
*that - it was released about the same time as the EMAX, and was a
*clever step between linear interpolation (which is still largely
*used today and has substantial aliasing artifacts) and a high-order
*interpolator.

*Let me know what else you conclude, and thanks for the brief
*interruption in my usual workday.

*Best Regards,
*--dave R
 
> Any schematic or anything? I have the IIx service manual
> but it conveniently omits the voice card schematic.

If you just wait a minute or two i will ask the original
Fairlight CMI III voice card designer how he designed them.

Regards
Kd....