> Did you foresee that people would actually treat your little
> VLSI-chip like an instrument?

Actually, I was an electronic music hobbyist before I started working for MOS Technology (one of Commodore's chip divisions at the time) and before I knew anything at all about VLSI chip design. One of the reasons I was hired was my knowledge of music synthesis was deemed valuable for future MOS/Commodore products. When I designed the SID chip, I was attempting to create a single-chip synthesizer voice which hopefully would find it's way into polyphonic/polytimbral synthesizers.

> Are you aware of the existence of programs like SIDPLAY,
> PlaySID,... which emulate the SID chip up to the smallest click ?

I only recently became aware of them (through your website).
I'm afraid I haven't thought much about SID in the last 15 years...I am constantly amazed and gratified at the number of people who have been positively affected by the SID chip and the Commodore 64 (which I also designed) and who continue to do productive things with them despite their "obsolescence".

> Have you heard the tunes by Rob Hubbard, Martin Galway, Tim
> Follin, Jeroen Tel, and all the other composers ?

I'm afraid not, are recordings available in the US?

> Did you believe this was possible to do with your chip?

Since I haven't heard them I'm not sure what we are talking about, however, I did design the SID chip with enough resolution to produce high-quality music. I was never able to refine the Signal-to-noise ratio to the level I wanted, though.

> How much of the architecture in the SID inspired you when
> working with the Ensoniq synthesizers?

The SID chip was my first attempt at a phase-accumulating oscillator, which is the heart of all wavetable synthesis systems.
Due to time constraints, the oscillators in SID were not multiplexed, therefore they took up a lot of chip area, constraining the number of voices I could fit on a chip. All ENSONIQ sound chips use a multiplexed oscillator which allows us to produce at least 32 voices per chip. Aside from that, little else of SID is to be found in our designs, which more closely resemble the Mountain Computer sound card for the Apple II (the basis of the Alpha Syntauri system). The DOC I chip (used in the Mirage and ESQ-1) was modeled on this sound card.
Our current designs, which include waveform interpolation, digital filters and digital effects are new designs that aren't really based on anything other than our imaginations.

> How big impact do you think the SID had on the synthesizer
> industry?

Well, I don't think it had much impact on the synthesizer industry. I remember once at Commodore that Sequential Circuits was interested in buying the chip, but nothing ever came of it. My intention in designing the chip (since MOS Technology was a merchant semiconductor house at the time and sold chips to the outside world) was to be able to sell the SID chip to synthesizer manufacturers. SID chip production was completely consumed by the Commodore 64 and by the time chips were readily available, I had left Commodore and never had the opportunity to improve the fidelity of the chip.

> What would you have changed in the SIDs design, if you had a
> bigger budget from Commodore ?

The issue wasn't budget, it was development time and chip size constraints. The design/prototype/debug/production schedule of the SID chip, VIC II chip and Commodore 64 were incredibly tight (some would say impossibly tight)--we did things faster than Commodore had ever done before and were never able to repeat after! If I had had more time, I would have developed a proper MOS op-amp which would have eliminated the signal leakage which occurred when the volume of the voice was supposed to be zero. This lead to poor signal-to-noise ratio, although it could be dealt with by stopping the oscillator. It would also have greatly improved the filter, particularly in achieving high resonance. I originally planned to have an exponential look-up table to provide a direct translation for the equal-tempered scale, but it took up too much silicon and it was easy enough to do in software anyway.

> The SID is very complex for its time. Why didn't you settle
> with an easier design ?

I thought the sound chips on the market (including those in the Atari computers) were primitive and obviously had been designed by people who knew nothing about music. As I said previously, I was attempting to create a synthesizer chip which could be used in professional synthesizers.

> Do you still own a C64 (or another SID-equipped computer) ?

Sure, I have a couple of them (including the portable), but I honestly haven't turned them on in years.

> Did Commodore ever plan to build an improved successor to the SID?

I don't know. After I left I don't think there was anyone there who knew enough about music synthesis to do much more than improve the yield of the SID chip. I would have liked to have improved the SID chip before we had to release to production, but I doubt it would have made any difference to the success of the Commodore 64.

> Can you give us a short overview of the SIDs internal architecture ?

It's pretty brute-force, I didn't have time to be elegant.
Each "voice" consisted of an Oscillator, a Waveform Generator, a Waveform Selector, a Waveform D/A converter, a Multiplying D/A converter for amplitude control and an Envelope Generator for modulation. The analog output of each voice could be sent through a Multimode Analog Filter or bypass the filter and a final Multiplying D/A converter provided overall manual volume control.
As I recall, the Oscillator is a 24-bit phase-accumulating design of which the lower 16-bits are programmable for pitch control. [1] The output of the accumulator goes directly to a D/A converter through a waveform selector. Normally, the output of a phase-accumulating oscillator would be used as an address into memory which contained a wavetable, but SID had to be entirely self-contained and there was no room at all for a wavetable on the chip.
The Sawtooth waveform was created by sending the upper 12-bits of the accumulator to the 12-bit Waveform D/A.
The Triangle waveform was created by using the MSB of the accumulator to invert the remaining upper 11 accumulator bits using EXOR gates. These 11 bits were then left-shifted (throwing away the MSB) and sent to the Waveform D/A (so the resolution of the triangle waveform was half that of the sawtooth, but the amplitude and frequency were the same).
The Pulse waveform was created by sending the upper 12-bits of the accumulator to a 12-bit digital comparator. The output of the comparator was either a one or a zero. This single output was then sent to all 12 bits of the Waveform D/A.
The Noise waveform was created using a 23-bit pseudo-random sequence generator (i.e., a shift register with specific outputs fed back to the input through combinatorial logic). [2] The shift register was clocked by one of the intermediate bits of the accumulator to keep the frequency content of the noise waveform relatively the same as the pitched waveforms. The upper 12-bits of the shift register were sent to the Waveform D/A.
Since all of the waveforms were just digital bits, the Waveform Selector consisted of multiplexers that selected which waveform bits would be sent to the Waveform D/A. The multiplexers were single transistors and did not provide a "lock-out", allowing combinations of the waveforms to be selected. The combination was actually a logical ANDing of the bits of each waveform, which produced unpredictable results, so I didn't encourage this, especially since it could lock up the pseudo-random sequence generator by filling it with zeroes. [3]
[Actually, the result isn't a logical ANDing at all. -TechEd]
The output of the Waveform D/A (which was an analog voltage at this point) was fed into the reference input of an 8-bit multiplying D/A, creating a DCA (digitally-controlled-amplifier). The digital control word which modulated the amplitude of the waveform came from the Envelope Generator.
The Envelope Generator was simply an 8-bit up/down counter which, when triggered by the Gate bit, counted from 0 to 255 at the Attack rate,

from 255 down to the programmed Sustain value at the Decay rate, remained at the Sustain value until the Gate bit was cleared then counted down from the Sustain value to 0 at the Release rate.
A programmable frequency divider was used to set the various rates (unfortunately I don't remember how many bits the divider was, either 12 or 16 bits). A small look-up table translated the 16 register-programmable values to the appropriate number to load into the frequency divider. Depending on what state the Envelope Generator was in (i.e. ADS or R), the appropriate register would be selected and that number would be translated and loaded into the divider. Obviously it would have been better to have individual bit control of the divider which would have provided great resolution for each rate, however I did not have enough silicon area for a lot of register bits.
Using this approach, I was able to cram a wide range of rates into 4 bits, allowing the ADSR to be defined in two bytes instead of eight. The actual numbers in the look-up table were arrived at subjectively by setting up typical patches on a Sequential Circuits Pro-1 and measuring the envelope times by ear (which is why the available rates seem strange)!
In order to more closely model the exponential decay of sounds, another look-up table on the output of the Envelope Generator would sequentially divide the clock to the Envelope Generator by two at specific counts in the Decay and Release cycles. This created a piece-wise linear approximation of an exponential. I was particularly happy how well this worked considering the simplicity of the circuitry. The Attack, however, was linear, but this sounded fine.
A digital comparator was used for the Sustain function. The upper four bits of the Up/Down counter were compared to the programmed Sustain value and would stop the clock to the Envelope Generator when the counter counted down to the Sustain value. This created 16 linearly spaced sustain levels without havingto go through a look-up table translation between the 4-bit register value and the 8-bit Envelope Generator output. It also meant that sustain levels were adjustable in steps of 16. Again, more register bits would have provided higher resolution.
When the Gate bit was cleared, the clock would again be enabled, allowing the counter to count down to zero. Like an analog envelope generator, the SID Envelope Generator would track the Sustain level if it was changed to a lower value during the Sustain portion of the envelope, however, it would not count UP if the Sustain level were set higher.
The 8-bit output of the Envelope Generator was then sent to the Multiplying D/A converter to modulate the amplitude of the selected Oscillator Waveform (to be technically accurate, actually the waveform was modulating the output of the Envelope Generator, but the result is the same).
Hard Sync was accomplished by clearing the accumulator of an Oscillator based on the accumulator MSB of the previous oscillator. [4]
Ring Modulation was accomplished by substituting the accumulator MSB of an oscillator in the EXOR function of the triangle waveform generator with the accumulator MSB of the previous oscillator. [5]
That is why the triangle waveform must be selected to use Ring Modulation.
The Filter was a classic multi-mode (state variable) VCF design. There was no way to create a variable transconductance amplifier in our NMOS process, so I simply used FETs as voltage-controlled resistors to control the cutoff frequency. An 11-bit D/A converter generates the control voltage for the FETs (it's actually a 12-bit D/A, but the LSB had no audible affect so I disconnected it!).
Filter resonance was controlled by a 4-bit weighted resistor ladder. Each bit would turn on one of the weighted resistors and allow a portion of the output to feed back to the input. The state-variable design provided simultaneous low-pass, band-pass and high-pass outputs. Analog switches selected which combination of outputs were sent to the final amplifier (a notch filter was created by enabling both the high and low-pass outputs simultaneously).
The filter is the worst part of SID because I could not create high-gain op-amps in NMOS, which were essential to a resonant filter. In addition, the resistance of the FETs varied considerably with processing, so different lots of SID chips had different cutoff frequency characteristics. I knew it wouldn't work very well, but it was better than nothing and I didn't have time to make it better.
Analog switches were also used to either route an Oscillator output through or around the filter to the final amplifier. The final amp was a 4-bit multiplying D/A converter which allowed the volume of the output signal to be controlled. By stopping an Oscillator, it was possible to apply a DC voltage to this D/A. Audio could then be created by having the microprocessor write the Final Volume register in real-time. Game programs often used this method to synthesize speech or play "sampled" sounds.
An external audio input could also be mixed in at the final amp or processed through the filter.
The Modulation registers were probably never used since they could easily be simulated in software without having to give up a voice. For novice programmers they provided a way to create vibrato or filter sweeps without having to write much code (just read the value from the modulation register and write it back to the frequency register).
These registers just give microprocessor access to the upper 8 bits of the instantaneous value of the waveform and envelope of Voice 3. Since you probably wouldn't want to hear the modulation source in the audio output, an analog switch was provided to turn off the audio output of Voice 3.

> Any other interesting tidbits or anecdotes ?

The funniest thing I remember was getting in a whole bunch of C-64 video games which had been written in Japan. The Japanese are so obsessed with technical specifications that they had written their code according to a SID spec. sheet (which I had written before SID prototypes even existed). Needless to say, the specs were not accurate. Rather than correct the obvious errors in their code, they produced games with out of tune sounds and filter settings that produced only quiet, muffled sound at the output. As far as they were concerned, it didn't matter that their code sounded all wrong, they had written their code correctly according to the spec. and that was all that mattered!

--

For questions or comments, Mr. Andreas Varga may be reached through his SID

Homepage on the World Wide Web at - http://stud1.tuwien.ac.at/~e9426444/

 Addendum by S. Judd, Technical Editor

------+-----+----------+-----+-------

References / suggested reading:

"Design case history: the Commodore 64", Tekal S. Perry and Paul Wallich, IEEE Specturm, March 1985

"Programming the Commodore 64", Raeto Colin West, Compute Publications.

"Mapping the Commodore 128", Otis Cowper, Compute Publications.

The SID Homepage, maintained by Andreas Varga:
http://stud1.tuwien.ac.at/~e9426444/

"SID Primer: The Working Man's Guide to SID" by Stephen L. Judd, disC=overy issue 2

"Mapping the Commodore 64", Sheldon Leemon, Compute Publications.

"Commodore 64/128 Programmer's Reference Guide", CBM.

The first article makes for very interesting reading and should be available at most public libraries. The second and third references have more detailed and accurate explanations of SID and the theory behind its general features, as well as actual implementation. The SID homepage has lots of technical and general information. The disC=overy article attempts to provide a general overview of the chip. The last two are included as good references for information on programming SID, especially since they are easy to acquire.

Notes:

[1] In the words of Andreas Boose:

"The phase accumulating oscillator is just a 24-bit accumulator which is increased by the 16-bit value of the frequency register every phi2 cycle. And like Bob said, the upper 12 bits of this accumulator are sent to the waveform generators.
Note that although he uses 12 bit, the resulting *resolution* of this signal is only 12 bit on lower frequencies, if the frequency register is smaller than 4096. On higher frequencies the resolution drops down to nearly 8-bit when the frequency register is at its max value (65535)."
It should now be clear why all of SID's waveforms are linear in nature (i.e. composed of straight lines): instead of using this counter as an index into a wave table, only the counter itself is used in generating the waveforms (and counting up is an awfully linear process). On the other hand, this raises the intriguing possibility of modulating the waveform via rapid changes in the frequency register.

[2] Asger Alstrup Nielsen has done a good deal of research into SID's random number generator for generating the noise waveform. I am told that the algorithm was implemented by Michael Schwendt in SIDplay and the result was not too accurate. For more information, visit the SID home page, in the references above.

[3] From the IEEE article referenced above:

The precise capabilities of the sound chip are not clear even today, largely because of incorrect specifications having been written when the chip was first designed. "The spec. sheet got distributed and copied and rewritten by various people until it made practically no sense anymore," said Yannes. An example of the faulty documentation is the claim that the chip can logically AND several waveforms. ...
"There is no interlock to make sure that if one bit is on, the others are off," Yannes said. "That would have taken too much silicon."
So if more than one waveform is selected, the internal nodes of the output multiplexer are discharged, and what emerges is the minimum of amplitudes."
The meaning of that last statement is unclear, as the result is certainly not the minimum of the waveform values either. A very simple way to see what the result looks like is to use the "wave3" program in the other disC=overy article with voice 3. To test the logical ANDing hypothesis, "freeze" the voice 3 pulse waveform output at $FF by selecting pulse and a nonzero pulse width. Select a very low note frequency, 1 or 2 say. When the waveform output  becomes $FF set the pulse width to zero: the pulse output is now stuck at $FF. Select another waveform, such as sawtooth, and watch (and listen to) the result! (To hear the result, make sure the sustain level is set (to 15 say) and that the gate bit is set as well). About the only thing that can be said about multiple waveforms is that they are periodic with the expected frequency.
In short, nobody really knows what the result is when multiple waveforms are selected!

[4] It should be clear that synchronization generates a new waveform at a fundamental frequency related to the preceding voice. Moreover, resetting the accumulator to zero will in general introduce a discontinuity into the waveform. Discontinuities will always amplify the presence of high frequencies, as can be seen by taking the Fourier transform of a discontinuous function. Compare for instance the triangle waveform, whose mode amplitudes decay as 1/k^2, and the sawtooth waveform, whose mode amplitudes decay as 1/k, where k is the wave number. High frequencies are needed to make sharp transitions.
Try synchronizing a triangle waveform -- if the synchronization frequencies are very different, the result will unsurprisingly sound like a sawtooth. The higher frequencies are easy to hear; sometimes even the fundamental pitch of the note will change significantly.

[5] Ring Modulation. Most books, not to mention the other SID article elsewhere in this issue of disC=overy, will explain ring modulation as a multiplication of two waveforms, which generates sum and difference frequencies. Simply put, if one signal is cos(w1*t) and another signal is cos(w2*t), then the ring modulated output is:

cos(w1*t) * cos(w2*t) = 1/2 (cos((w1+w2)*t) + cos((w1-w2)*t) )

Voila! Two new waves, with frequencies which are the sum and difference of the two frequencies. Note that if the two frequencies differ only slightly, very noticable beats will result.
SID works a little differently. What it does not do is to multiply any waveforms. What it does do is generate a new waveform, somewhat related to the original waveform, which contains all the sum and difference frequencies.
Recall that the triangle waveform may be written essentially as:

f(x) = { x, MSB=0 i.e. 0 < x <= max/2

{ max-x, MSB=1 i.e. max/2 < x < max

where max is the maximum value of the accumulator, 2^24. f(x) then counts upwards on one half of the cycle and downwards on the other half.
Ring modulation uses the MSB of the preceeding voice to evaluate the function above. The normal triangle waveform is continuous, since when x=max/2 the two pieces of the function have the same value. If MSB is suddenly changed, however, the function, which was couting upwards, might suddenly change values significantly and start counting downwards. It is pretty simple to sketch out the result on a piece of paper. Consider the following:

V3: At some frequency a little higher than V1 (waveform doesn't matter)

V1: A triangle wave at some base frequency, modulated by V3.

Modulated output of V1:

------------*-------*[--------------------------------- maximum value
[* * [
[ * * [
* [ * [ [
* [ [ * [ [ * etc.
* [ [ * [ *
* [ [ [ *
* * [ [ *
* * [ [ *
* * [ [ *
*----------*[--------[*-------------------------------- minimum value
^ ^ ^ ^
[ [ [ [
[ [ [ Curve hits FF and wraps around to 00
[ [ [
[ [ V3 completes a cycle; MSB is cleared; V1 wave counts up again
[ [
[ Curve hits FF and wraps around to 00; MSB is high so result
[ is 255-value (i.e. counting downwards)
[

V3 hits midway point in cycle; MSB is now high; V1 goes from t to 255-t, goes downwards.

Well it's not really 0..255, but the meaning should be clear. :)

There are now two important features: first, a number of discontinuities have been suddenly introduced, which tends to amplify higher frequencies. The second is that the frequency of the resulting waveform is quite different than that of the pure triangle. What is the frequency? To put it another way, what is the period: how long must one wait before the waveform repeats itself?
In terms of SID, let both of the accumulators start at zero, and let one count up faster than the other. The period of a given voice is the amount of time it takes for its counter to return to zero. At some point both counters will return to zero, and of course the modulated wave will then repeat. This then gives the period of the modulated waveform.
Let one wave have period T1 (frequency w1) and the other have period T3 (and frequency w3). When m*T1 = n*T3, for some integers m and n, the wave will repeat. That is, perhaps after ten repetitions of wave 1 and three repetitions of wave 3 the waves will both be at zero again. The resulting wave will then have frequencies w=(p*w1 + q*w3) for integer p,q. To see this, consider a wave like:

f(t) = e^i 2*pi (p*w1 + q*w3) t, where w1=1/T1, w3=1/T3

let t = T = m*T1 = n*T3, then

f(T) = e^i 2*pi (p*m + q*n) = f(2*pi) = f(0)

Thus, such a wave has period T (although not necessarily least period T), for *any* p,q integer. This then implies that the resulting waveform contains all of the sum and difference frequencies, e.g. w1-w3, w1+w3, 2*w1+w3, etc.

Q.E.D.

In summary, SID's ring modulation produces a new waveform containing sum and difference frequencies of the modulating waveforms, with a strong high frequency content due to the discontinuities in the resulting modulated waveform.
Note that by using ring modulation (not to mention sync) frequencies much much higher than than the upper frequency limit (~4000 Hz) of the frequency register may be generated.