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My pleasure. Btw. is there some demo song, showing the SquatE abilities with all bells and whistles? Especially, that non-interupting samples must be great.

Definitely. For background chords in the Amiga minor/major style. But still, I think that vocals are even cheaper to play. When 32 samples are enough on GB, are 16 probably as well... and this way, the looping and separation could be trivialised, probably. It's also of little importance, whether the voice sounds harsh, distorted by quantisation - one could take it rather as a feature... we all love robotic voices, especially when they are in-tune .

Great list. Everything there was new for me, nice to see such huge advances in this techinque.

Do you know LPC fundamentals and this marvelous document? https://cnx.org/contents/swFM2W46@5.12: … troduction
It describes details of development of famous 80s SpeakNSpell toy - first LPC voice synthesizer chip. I think while LPC is fully computable on ZX (you can nicely interleave each LPC value processing with pulse width value decrement etc.), microsamples are better suited for this. We really need just few of them, they are ultrashort, might be 1-bit, and there's no computational overhead at all...

Very impressive research, nice sound! I think the 1bit sound is more advanced than AY today (no surprise). And because there's no reliable ratio between CPU and AY clock, and AY register access is clumsy, there's no need to bother with this PSG... or is?

Has somebody  experimented with volume registers on AY and some kind of advanced 1-bit mixing? The logarithmic nature of volume curve might help solve some membrane-pressure sideeffects...

Z.

finally!

this concept is actually pretty cheap (you can "emulate" low pass over the buffer just with shifting the window over longer sample content or by switching more buffers in a banked manner).

sounds fucking great, congratulations, utz! 1 bit is enough!

Z.

Well, the key idea is, why do all the people use 8bit counters (DEC counter8; JR NZ 10T_padding; LD counter,divider; LD channel,spike) - this means the higher tone, the less accurate its frequency (divider has just few valid bits). this takes 26T/channel AFAIK.

But, there's different approach - phase accumulator instead of divider, there's no need for conditional jumps, you just add to 8bit "accumulator", ala
LD A,frequency; ADD A,L; LD L,A (no need to reload counters, carry is immediately available, you even can use ADD HL,DE to accomplish this, when the frequency of two channels is very different - bass carry-contributions from L to H could be neglected then etc.).

Carry then feeds (SBC A,A; AND scaling_val; ADD A,scaler8) another register, "scaler", where's accumulated scaling value, typically some power-of-two, repeatedly added and we can control octave this wat, ensuring that the frequency would be in perfect align with other octaves, because we add 128, 64, 32 or similar "nice" numbers. And, when the numbers are even a little bit biased, let's say by 7, 5 or so, then, PWM control degrades for very low widhts to "intermittent" pins, giving nice fades. You could even produce harmonics with "masking" of that scaler, in one channel.

You can think of it like a <#80-#FF>-range phase accumulator as "top-octave" generator, with secondary 8bit scaler fed by its pins, doing the "downscaling" stuff - volumes, timbre, intermittent PWM stuff.

I have this idea for more than 10 years, but still no time to implement it properly (some proof-of-concept has been made on PIC12F628 :-), where it takes 7 mcycles (7us for 4MHz xtal) per channel...

regards,
Zilogat0r

Interesting reading, guys! And a great reverse-engineering attempt. But, just few notes to it:

- why is the loop djnzed, not unrolled, etc? it's because there's no need for extra speed, slow iteration means longer spike (more audible pin-based channels), and it was originally written with code-size in my mind, for Hellboj^HPRG (who loved Hubbard's JSW so much that i decided he deserves this gift:). summer 2000...

- channel counter preset - it helps to fight stuck-one in the bass channel (resting pin channel would never produce this), but more importantnly, it helps fight the aliasing - imagine playing two tones simultaneously, just one octave apart. pre-setting channels with different starting values (0,1,2 or maybe 0, 2, 4?) will produce desired harmonics, otherwise spikes of the higher tone would cover up the spikes of the lower tone completely. it's audible in the JSW song in one pattern - data are correctly there, but the original routine (without preset AFAIK) plays like there's no progression in one of the channels - all the spikes for that particular tone were just aliased out.

Now, i'll play with your converter, Utz!

Z.