Spin semiconductor fv-1

[puretube]

err, thank You for that kind offer, Jack; 

but I`m not intending to go into the depth of programming
this thing...
(but I had me sent the quotes from the euro-distributor,
who btw doesn`t have the "Spin" on his website, yet...).

[MR COFFEE]

Hi Jack,

Have you tried using it to develop envelope signals to control analog functions or filters (using the DSP digital filter capabilities or off the chip in analog using a DAC as a CV output)?

I'm still reading through the instruction set (assembler manual) and haven't got a feel for the capabilities of the chip yet beyond delay, reverb, and chorus\flange effects. Perhaps reading it all and looking through the code examples from the presets will make things clearer to me, and I just need more time with it. I'm trying to decide if I want to go for the development kit.

I've never programmed DSPs, but I have done assembler on plain vanilla uPs, and I'm trying to figure out if I can learn to program the thing in a reasonable amount of time. Barr writes in "his corner" that it is supposed to be fairly easy to program even for analog\non-digital types. Do you think the FV-1 seems harder to you because the instruction set is just so different from the general purpose DSP (so that someone like yourself who is used to the general purpose DSP instruction set has to stop and rethink how to do the same things on the FV-1 in a totally different way determined by the limited instruction set, i.e., do it Keith's way), or do you think the FV-1 instruction set is inherently flawed\awkward\too limited for anything except delay, reverb, and chorus\flange effects? I'm debating getting a development kit, but I don't want to blow $130 bucks to find out I can't really do what I want on it...

I'd like to hear more about your experience with it if you have time to share your thoughts about it.

Thanks

[MetalGuy]

What is the current situation - how/where can one get any?

[puretube]

reduce the link in the 1st post to the basic URL,
and then browse the page thoroughly...

[Peter Snowberg]

The FV-1 is fresh out of the gate and I don't know if all the final testing has been done yet. They should be available soon for $10 each in quantity 1-99 from OCT Distribution, Spin's U.S. distributor.

Maybe we can get Small Bear to pick this chip up? (are you listening Steve?)

I have a pair of them and THIS IS A REALLY EXCITING CHIP! I also got a nice e-mail today from the FV-1's designer, Keith Barr (who was also the founder of Alesis and MXR). He even came through here (diystompboxes.com) today.

I've got four FV-1 circuits on paper so far, including one that will be posted here for the DIY crowd under the name Lotus. It's an eight effect, four chip design including the voltage regulator . As this does not require the builder to program a microcontroller, this circuit should be accessable to a pretty wide audience.

[MetalGuy]

It's an eight effect, four chip design

Do you mean 4 chips one after another? Something like chorus-delay-reverb chain or something?

[puretube]

i guess:
1.: power regulator
2.: input "conditioner" (buffer & splitter)
3.: FV-1 ("jack-of-all-trades)
4.: output "summer" (mixer)

[Peter Snowberg]

Very close . The four chips are...

1. Power regulator to provide the logic with 3.3V (URL)
2. Input "conditioner" (buffer & splitter) & output mixer (TL072)
3. FV-1 (jack-of-all-trades)
4. (optional) E²PROM for adding/changing algorithms (24LC32)

To this list, add a crystal and a handfull of resistors and caps, and you have a complete DSP effects unit with 1 second of delay memory.

I think most of the fun around here will be from changing the contents of the E²PROM. This gets us into the arena of I.P. protection as the real "work" becomes a series of easily accessible bytes in a generic memory chip. To head off any issues, all the code for the Lotus will be OPEN SOURCE. 

There is an extention to the Lotus design that adds a little microcontroller, a USB bridge chip, and a pair of optoisolators for programming while remaining free from any possible ground loops. The microcontroller serves as a translator to move serial data embedded in USB packets into I²C packets for writing into the E²PROM.

If you don't want to program effects but you want to change the operation of the device, all it takes is replacing the 24LC32 with a new chip which has been pre-programmed with different effects code.

Now, anybody want to guess what the six controls on the top of the box will be?

[puretube]

hehe: just layouting an SMD PCB, with one conventional thru-hole part: a DIP8...

[Peter Snowberg]

hehe: just layouting an SMD PCB, with one conventional thru-hole part: a DIP8...

[DaveTV]

Okay, Peter, I'll give your control question a shot.

Controls 1, 2, and 3 are the 50k pots used as input controls
Control 4 is a blend knob for mixing the dry and wet signal
Control 5 is a program selector switch to switch between the 8 stored effects on the EEPROM (but not the internal effects, since it sounds like you're not using those).
Control 6 is a...gain knob? If not, it'll be a knob labeled "Good" that does just that.

The FV-1 does look like an interesting chip. After spending a lot time with the AL3201B, I think I'm ready to try something new.

[PharaohAmps]

Ordered my dev kit this afternoon.  Stay tuned...

Matt Farrow

[puretube]

I`ll give it a closer look with a magnifying glass, once they arrived...

so the mailman just rang the bell:

AL3201 & CS4271... driven by a 8.192 quartz
plus: 2115 & a tiny ball-grid-array chippy I can`t decipher... (eprom?/µP?).

(btw: there`s a tiny name printed on the PCB beneath the word "NemFX": "Marshall"

[PharaohAmps]

Just got the FV-1 eval board.  Actually, I got it about 5 hours ago, and I've been playing with it all evening.  So far I've "massaged" the delay example to suit my purposes and it seems to be going well.  I'm definitely not used to assembler - only 1 variable!  Well, not really, but the whole wrax, rdax thing is a bit new to me.

Anyway, sound quality is excellent, though the eval board is mostly set up to be used through a mixer.  I stuck a simple opamp buffer / mixer on the little dev area on the eval board, and it's working fine for me so far.  I'll keep everyone posted as I mangle this thing up some more.

Matt Farrow

[smallbearelec]

The FV-1 is fresh out of the gate and I don't know if all the final testing has been done yet. They should be available soon for $10 each in quantity 1-99 from OCT Distribution, Spin's U.S. distributor.

Maybe we can get Small Bear to pick this chip up? (are you listening Steve?)

I will contact them as soon as I can get out from under stuff backed up from the trip.

[PharaohAmps]

BTW, the guy from OCT quoted me $8.50 for these chips in 100-1000s qty.  If you just want to use the ROM programs in mono, you can build it with 1x dual opamp, the FV-1, and a 3.3V regulator.  The ROM programs ARE pretty good, as you'd expect from Keith Barr.  There's some flutter in the pitch-shift algorithm, but the reverbs are really nice, and the flange and chorus are pretty decent.  There's some really smart code in the chorus program that scales the width of the chorus as the rate changes.  Lots of good info there.

I've got the example delay program outputting the straight signal as well as using one of the pots to control the delay signal, and it looks like you can easily run this thing in stereo with just a dual op-amp to buffer the input.  The output is beefy enough to be used as-is.  However, this loses you one of the onboard pots since you have to use it as a mixer.  But it does mean that you can build the thing all in software, which is nice.

I'm going to be messing around with some single-supply, low power opamps, so I can run the whole board from 3.3V.  So far the LMV324 quad looks real good, in a 14-pin SOIC.  I've got LM340-3.3's in SMD, as well as 33khz crystals.  The EEPROM is going to be DIP so I can program the thing in the eval board, but for what I want to do with it right now I can basically use the layout on the eval board, just slap a pair of buffers on there for stereo inputs.

Any of you DSP-heads out there have any idea about reverse delay using this thing?  I can write to the delay RAM, and then use a pot to control the pointer where the delay read happens.  I'm wondering if there is any way to read the samples out of the delay in reverse order...

Right now, I'm working on a through-zero flange by delaying the straight signal by 10mS, then running the flange delay from 2-20mS. 

Matt Farrow

[Peter Snowberg]

Dave:

You got it.   
Control 6 is output volume.

Steve:



I hope you're having a great time in South America!

Matt:

Very cool! I can't wait to hear your results!

[Mark Hammer]

Right now, I'm working on a through-zero flange by delaying the straight signal by 10mS, then running the flange delay from 2-20mS. 
Matt Farrow

Hi Matt,

You may want to scale back the fixed delay a bit.  Even though it probably rocks for post-production processing, 10msec seems to me to be rather unsuitable fixed delay for live performance.  The one advantage it presents is that one can spend more time on "the other side of zero" before returning.  On the other hand, that can also be achieved by creative use of sweep waveform.  Scott Stites (diyfreque) is probably the person to talk to here about what works and doesn't work well in TZF.

We've had numerous discussions on the main forum about TZF, and with so few people in a position to study it parametrically, it is difficult to reach some consensus about what is the minimum amount of fixed delay needed to achieve the most desirable effects.  Scott's site (forgot the name/URL, but I'm sure you'll be able to find it) has some nice illustrations of what we've come to call symmetrical and asymmetrical TZF.

[PharaohAmps]

You may want to scale back the fixed delay a bit.  Even though it probably rocks for post-production processing, 10msec seems to me to be rather unsuitable fixed delay for live performance.  The one advantage it presents is that one can spend more time on "the other side of zero" before returning.  On the other hand, that can also be achieved by creative use of sweep waveform.  Scott Stites (diyfreque) is probably the person to talk to here about what works and doesn't work well in TZF.

I remember the discussion about his Dimension C clone, I'll have to dig that back up.  Right now my test program is running with the fixed delay at 1310 samples, about 4 ms at 32.7khz clock rate.  The 10ms delay was definitely too much, but the experiments I did at 320 samples (1 ms) didn't get me enough time at zero.

Also, what you've said about LFO shaping is certainly relevant.  I'm now messing with the ramp LFO's to get an asymmetrical triangle wave so I can do more "swooping" about the zero point.  Right now it sounds good but lacks the drama I was looking for.

Next up is "barberpole" flanging, which from what I've read needs to incorporate a small amount of pitch shift.  Hard to do well in analog, but easy as pie with the FV-1.

Matt Farrow

[puretube]

anyone hearing (intermodulation) artefacts with the fading signal?

or is that just my sloppy (partly unshielded) input-wiring
picking up signals from 2 battleing
cheap switch-mode powersupplies
(one for the e-board, the other for the amp)?