How do they work? - echolevel/open-amiga-sampler GitHub Wiki

The way these samplers work is no great mystery, and it's precisely because their manufacture was so mercilessly low-cost that they're relatively easy to understand. Most of them contain a small PCB with an 8bit AD (analogue to digital) converter chip, a switching chip that's controlled by voltage (usually a quad bilateral switch), a couple of resistors and capacitors, and a 25-pin parallel port connector. That's all! The AD chip is 8bit/mono and capable of being driven at fairly high speeds - considerably faster than most Amigas would be able to drive it.

If the chip is mono, wouldn't you need two for stereo? Nope, that's what the switch is for: first read a sample byte from the left channel, then use the switch to send the right channel to the AD, read another sample, switch back, and so on. Sounds like an exercise in cost cutting, and it is, but since the Amiga's parallel port can only read 8 bits of data at a time, the luxury of dual AD chips wouldn't actually help (that's why some samplers got higher bitrates by simultaneously sending extra bits of data through the serial or joystick ports and adding them together in software to get e.g. 12, 14 or 16bit audio).

We'll not concern ourselves here with how the AD chip converts analogue audio input into 8bit digital output (if you're interested, check out some datasheets for chips like the ZN449, the AD7576 or the ADC0820 which detail their internal circuitry). We're more interested in how to control these chips - to instruct them to start a conversion (a 'sample' of the incoming audio), write the resulting byte to its output data pins, and then do it all over again really fast: as many times per second as our desired sample rate. Oh, and specifically, how to do that with an Amiga!

As you know, the Amiga is the best at everything. No surprise, then, that it has the best parallel port spec! (Actually the Amiga 1000's parallel port was a bit weird, but that's another story.) It's very easy to use, and very flexible due to the fact that its 8 data lines and some of its control lines are bidirectional. A very simplified explanation of how it's used for sampling is as follows:

• Our Amiga (ASM, C, etc) code performs a read of the parallel port's data byte at the relevant hardware address
• The Amiga's CIA chip automatically fires a STROBE signal on Pin 1 every time a data byte is read

That's it! But that's no good - we've only got one sample byte, and it's either empty or gibberish because the sampler hadn't been instructed to perform an analogue-to-digital conversion. That's what the STROBE signal instructs it to do. It performed the conversion AFTER we read the byte. So:

• We set up a loop in our code, using an interrupt to set the speed of this loop - and the speed of the loop should be our desired sample rate
• We come to terms with the fact that the first data byte we read will always be useless - perhaps we don't mind, or perhaps we chop it off later, it doesn't matter
• That first data byte read will trigger the first STROBE, and then the loop of "read byte from parallel pins -> send STROBE -> trigger new conversion on AD chip -> AD chip sends byte to parallel pins" continues until we break it (often with a mouse click, or automatically if we run out of memory to store all these bytes)
• Usually we'll send every incoming byte to Paula, the Amiga's audio chip, to be played back from the Amiga's audio output. This means we can monitor the incoming signal even if we're not storing the data anywhere - useful to decide if we need to tweak the input level, wait for just the right moment to start recording, or whatever.

So the bare minimum the Amiga needs to do is to read bytes and send STROBE pulses at some arbitrary fixed rate. That rate, according to the Nyquist–Shannon sampling theorem, should be greater than double the highest frequency you want human ears to be able to hear, but it also needs to be lower than the highest sample rate the Amiga is capable of playing back - so between about 8Khz and 40Khz is what most Amiga musicians will use in normal circumstances.

Some Amiga software lets you use a slider to choose an arbitrary sample rate, whereas Protracker samples at the 'note' of your choice between C-1 and B-3. The note references a lookup table of sample periods, which divide e.g. a PAL Amiga's clock rate of 3546895hz (NOT the processor speed which varies from model to model) to calculate an interrupt frequency that serves as the sample rate. For example, the note A-1 with a finetune offset of 0 has a period of 508, so 3546895/508 = 6982.07Hz, so if you want a sample rate of about 7Khz, you sample at Protracker note A-1. YES, THAT'S WEIRD. Protracker's weird. Amazing and beautiful and weird. The external sound source that you sample at 'note A-1' doesn't have to be a real-world A-1 note (which would have a frequency of 55Hz, as it happens). It could be anything. So assuming your sound source is playing at the pitch you want to record, the PT note at which you sample is just a metric of sample rate and a way of deciding your preferred quality-to-filesize ratio. Oh, and remember I mentioned PAL? That clock rate is different on NTSC machines which means that, you guessed it, pitches are all slightly different If all this makes sort of but not complete sense, welcome to Protracker!