Timeline for How did musicians acquire samples for tracker music (MOD, S3M, XM and the like)?
Current License: CC BY-SA 4.0
9 events
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| Aug 7, 2020 at 21:08 | comment | added | supercat | @Graham: Alternatively, one could perhaps plug into the user port instead of the cassette port, and exploit one of the CIA's CNT inputs to sample the input on every other clock cycle. Let the counter free-run and simply read it periodically, and have each sample be a weighted average of different readings. | |
| Aug 7, 2020 at 20:41 | comment | added | supercat |
@Graham: If one uses a crude analog comparator to check whether the SID output is above or below midpoint, and samples that in an unrolled loop cpx 1 / adc #0, I would expect that such a design would provide a certain amount of anti-aliasing, which could be enhanced if one can store sums at 3x the desired sample rate and then does some simple filtering when down-sampling.
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| Aug 7, 2020 at 20:10 | comment | added | Graham | @supercat And by the way, it's also why the SID did such a good electric guitar sound. Most of the higher frequencies on an electric guitar are enharmonic artifacts from distortion, so adding a bunch of potent sources of enharmonic noise is almost a good thing! It's less good for speech and anything like piano which needs to be reproduced more accurately, of course. | |
| Aug 7, 2020 at 20:05 | comment | added | Graham | @supercat The problem is the gap between the Nyquist limit (half sampling rate) and the frequencies you want to keep. The smaller that gap, the harder your filtering needs to be, but harder filtering also screws up the phase which isn't good either. It was digital filtering (in hardware) which really sorted things for ADCs for the tiny gap between 20kHz audio and 22kHz Nyquist. It's also the benefit of 88/96kHz sampling; the sample rate doesn't improve the data, it improves the anti-aliasing. | |
| Aug 7, 2020 at 17:07 | comment | added | supercat | ...4-bit 8kHz quality audio. Program the SID to take the audio input, filter it, mix it with a noise waveform, and output it. Then use the transistor to detect whether the output is above or below a desired threshold. Maybe use two transistors if the response from using one would be too mushy. | |
| Aug 7, 2020 at 17:05 | comment | added | supercat | I find it somewhat curious that audio capture hardware was obscure enough that programs to play clips of digitized music were popular downloads from services like Quantum Link, even though the circuitry required to capture audio at the quality of such files would have been sufficiently simple and cheap that the cost of the connectors could have exceeded the cost of all the other components combined. Though now I'm almost feeling inspired to try building something. For use with a C64 or C128, the hardware budget might have been reducible to a transistor and some resistors, for... | |
| Aug 6, 2020 at 14:42 | comment | added | DmytroL | Thanks @Graham! It is particularly educating that ADCs (even if 8-bit ones) were commonly available in the 80s so it was economically feasible to use them in consumer hardware. | |
| Aug 6, 2020 at 10:24 | history | edited | Graham | CC BY-SA 4.0 |
added 277 characters in body
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| Aug 6, 2020 at 10:16 | history | answered | Graham | CC BY-SA 4.0 |