More than 3 pots possible?

[livingston]

I was wondering if there's any way to kludge more than 3 pots into the FV-1? Seems like, at the very least, you could use one of the audio inputs to take a control voltage in, right?

I wanted to make a sort of pitch shift arpeggiator, with a knob for each of the 8 steps of a sequencer, plus a speed knob. Are there any workarounds that would make this possible?

[frank]

The ADC inputs really expect a DC free audio band signal, so using them as a POT input with a DC signal really won't work. You could generate an AC signal and use the amplitude as the control signal then just filter it in the FV-1 to recover the amplitude as a DC signal.

[livingston]

Is it possible to do something with multiplexing to get more pot inputs? I'm pretty new to digital hardware so I don't really know what would be involved.

[frank]

Nope, the FV-1 only supports 3 pots. While there are tricks that could be played with pot position, that causes a whole set of programming problems that are not worth the headache.

[wurgle]

Hi all

Sorry I'm a real newby so if this is a load of tosh please don't kick me too hard.

Most music does not need to go below 20Hz, perhaps even for say a guitar 40Hz.

So if we make a mixer on the audio input we can make some oscillators at e.g. 10Hz on the audio input and use the number cruncher to seperate it out.

LPF the inputs to extract the signal.

Use the FV-1 itself to make the oscilators on its output mixed with the audio and filter them.

That buys you at least 2 more pots without loosing an input. You could prob use 10Hz and 20Hz on each input to give you 4 pots more.

[livingston]

I guess the question would be: at what frequency does the chip's internal highpass filter begin? Could you even get a 10hz signal into the input?

You might also have some issue with banging against the supply rails, I would think. If you have a 10hz signal and your audio signal, I think you very well might clip the audio signal during the extremes of the 10hz wave.

[frank]

On top of what livingston mentioned, you have to consider the filtering to remove the 10Hz signal from the audio. For a single pole low pass filter you will drop 6db/octave. 40Hz is only 2 octaves away so a full scale signal, along with possible clipping as mentioned above, will only be 12db down on the output so a nice low frequency hum out the speakers.

A much heavier filter required to remove it, say 4 poles. Still not great as 4 poles will drop at 24db/octave so 2 octaves away means it is only 48db down at 40Hz.

[livingston]

I guess the question would be: at what frequency does the chip's internal highpass filter begin? Could you even get a 10hz signal into the input?

What do you think about this, Frank? I don't think it's a great idea to put a control signal on the same input as audio, but it would be useful to know what the low frequency limit of the inputs is. Do you have any data on that?

[frank]

I would need to check the exact roll off with Keith but we designed to be roughly 20-20K at 48KHz sample rate so really low frequencies will get rolled off. We don't test for below that so the response below 20Hz could vary over time as the process shifts at the fab.

[livingston]

Would the low frequency response be changed using a 32k crystal?

[frank]

Probably slightly, I would need to check a few things but the low end will be a factor of some fixed analog components and the clock rate since it is delta-sigma but it is not going to be a big change.

[donstavely]

For what it is worth, here is what I would do for your appegiator application:

First, I would use switches rather than pots for the eight steps. This may be to limiting for you, but it is pretty common.

Second, let's assume that you only need one signal input and one output, so the second FV-1 input and output can be used for the additional control signals.

Then I would wire eight SPST switches with power-of-two resistor values to create a switch-based D-to-A converter. This is what goes into the second signal input.

Since the signal input is AC-coupled, the switch DAC needs to be driven by an AC signal. You can use the second signal output driven by a two-integrator sine oscillator going almost rail-to-rail.

You then need to do the conversion back to digital codes in firmware - a little messy, but doable with eight subtracts and skips.

The two MSB resistors need to be better than 1%, due to the 8-bit accuracy.

You don't have pots on each arp step, but you do have all three pots left available. What if you used one to select which note to emphasize, and the other for emphasis strength.

Let me know if this is helpful.

[livingston]

That's interesting Don, but I don't exactly understand how it would be used. What would the switches control?

[donstavely]

What I was proposing was switches to enable or disable steps in the arpeggio sequence, like on many synths. Then you could have a pot-skip routine to select the arpeggio pattern (ascending, descending, up/down) and for the intervals(scales, chords, etc.) This alternative would not give you the ability to control the interval for each note in the pattern, as you could if you had a pot for each note.

BTW, you might find this interesting:
http://www.youtube.com/watch?v=e6idBRbO7RU


I was originally considering the Line6 dev kit. It was pretty cool, but the tools support was very poor. I don't think they got much traction.

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