Amplitude modulation affects the volume of the waveform. It is often used to produce vibrato or tremolo effects. Frequency modulation affects the period of the waveform. Although the basic waveform itself remains the same, the pitch is increased or decreased by frequency modulation.
Actually what you've described is a means to produce vibrato - rapidly modulating the pitch.
The whole idea of FM synthesis is to make the wavefrom itself change. It's the same thing as pitch modulation but applied on much smaller time scales. In the case of the Amiga you would have to alter the sample rate for every sample.
The real Amiga audio system by default was designed for playing samples. Contrast this to the C64's SID chip that used Attack, Decay, Sustain, Release circuits (hyrbid analog-digital). Both have advantages. I advocated adding ADSR circuits to the Amiga audio. But oh well. The Amiga does have a mode that goes beyond sample playing however. See Karlos's posting above.
SID used analogue oscillators and an alalogue filter so yes it's a hybrid technology. I'm surprised it wasn't used in synths at the time - it is now!
As for ADSR you don't need hardware in the Amiga for that, you can calculate the evvelope in software and write the values into the volume register.
>To properly emulate the Amiga sound system *accurately*
>will rather ironically, require some pretty high end (read expensive) kit.
Same is true of the video hardware. Though the Amiga videographic circuits are relatively incomplete by some modern standards, most graphics chips today lack sprites (it's true), color compression modes... and yes, most lack multiple playfields like the Amiga had. Instead they rely on brute force and fancy blitters. Sad but true.
I reckon Cell could do a killer chipset emulation.
As for the audio the reaon I'm saying it'll be difficult is because if you rely on using different sample rates for playback they'll introduce distortion at diffferent frequencies and this distortion is part of the sound. This wont make much of a difference at high smple rates but you'd be surprised how many mods used 8.3 KHz samples - and played them at lower rates.
At low rates the distrotion is going to be an integral part of the sound so has to be recreated to be accurate. Even these low sample rates wont "fit" properly into 44KHz so you'll need a sampling rate a lot higher to generate even accurate-ish waveforms.
Another way would be to use additive systhesis to add in the distortion, this might actually be *easier* but will require per sample calculation of the correct noise levels / frequency.
Then of course any inaccuracies in the DACs have to be modelled as well as the impact of later analogue components in the audio O/P system.
You may think that you don't want the distortion but it's an integral part of the sound and you'll most likely actually prefer it included.
The early Fairlight CMI samplers were only 8 bit but the users actually preffered their sound to the later 16 bit models.
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Hmm: I don't think I want to go into this level of complexity but this is giving me ideas for a new kind of Oscillator to add to my synth - I think I may a "Retro Sampler" module to the list...
I'm thinking of using 192KHz as the sampling rate anyway so should be able to do different sample playback rates fairly well.
