freddy
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Posts posted by freddy
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Or another way of looking at it, this code will set Port A 7 high for exactly .1 µs.
bsf PORTA, 7 ; set pin high bcf PORTA, 7 ; set pin low
:thumbsup:
This is not direct port setting but using SRIO, so I suppose the time will vary in my case. But knowing that NOP will take 1/10th of usec definitely helps.
Thanks again.
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nILS,
The PIC runs 10,000,000 instructions per second, that's 10,000 instructions per ms or 10 instructions per µs. If you want to wait for 1us you'll have to do nothing for 10 instructions. This can be easily done by simply doing NOP ten times.
bsf PORTA, 7 ; set pin high nop ; start waiting nop ; 2 nop ; 3 nop ; 4 nop ; 5 nop ; 6 nop ; 7 nop ; 8 nop ; 9 nop ; 10 bcf PORTA, 7 ; set pin low
The code above will set Port A 7 high for exactly 1us. If you wrap that into a macro it'll look much nicer ;)
yeah, dirty, but works. It's supposed to be a real delay so I don't really care about ineffectiveness of the code and see no reason to mess with interrupts for this. I'll make a label with a counter allowing me to set different delays.
Thanks for the hint! :rolleyes:
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Hello,
during programming of digipot chips connected to DOUT pins I have the need to trigger CLK signal for a particular time. Calling 1ms MIOS_Delay is much too long (updating 16 digitpots with 6 registers each with 11 bit words just takes much too long), using no delay at all (i.e. triggering the pin to 1 followed by 0) is much too fast for my logic analyzer. How to achieve e.g. 1us delay (doesn't need much too precise, +/- 100ns is more than enough)?
I cannot imagime how this might work with timer and within an interrupt-serve routine. I've tried to look through forums but couldn't find anything that might help me.
Thanks,
freddy
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I've abandoned the single-connector design in order to shorten audio paths and avoid parallel tracks. The current design is as shown in the attachment, I'll print it tomorrow and try building a prototype if no significant comments appear.
Tried it today, but whatever I do the output still looks the same. Regardless of which PCB design I use there is always 400kHz sine wave generated on inverting pin of IC1 (see attached schematics). The 400kHz frequency is determined by the oscilloscope. The sine wave and it's frequency stays the same regardless of input signal frequency and shape (using function generator, 50Hz-20kHz input wave, sine, triangle or square). This "noise wave" is then mixed into output, obviously, and causing noise as shown on previous photos.
Since the frequency stays the same regardless what I do I'd expect it to be some "environment" noise or error (though of unknown origin). I ran out of 100nF filter caps at the moment, so I'll finish building the gyrators later this week. Meanwhile I'll solder one AD5206 and depending ICs and start modifying MBMixer code to work with AD5206.
If anyone op-amp skilled looks at the schematics and an obvious reason for the noise occurs to him, please let me know as well :rolleyes:
The up-to-date EQ PCB design is attached for reference as well.
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I've abandoned the single-connector design in order to shorten audio paths and avoid parallel tracks. The current design is as shown in the attachment, I'll print it tomorrow and try building a prototype if no significant comments appear.
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Lyle,
thanks for comments, notes!
You should have one (or more) bypass capacitors at each chips power pins. These should be as close to the chip as possible.
This is true for digital circuits too, but even more important for analog stuff.
I wanted to place them directly on the IC pins on the copper layer, but right, they're not found on the PCB.
Try to avoid long parallel traces.. this can lead to "crosstalk" between audio signals, or even worse, from power leads to signal leads.
I doubt these can be avoided in case of a single 6-pot connector, but I've placed ground plane between each two signals. There are still power leads really close to IC pins, I hope these will not cause much damage.
Ample grounding is important... as is shielding, but don't use the same traces for both. Shielding traces should be carrying no current.
The ground arrangement in the last version of the mixer had four separate ground circuits! The analog ground, for all op-amp circuit grounding
and the analog part of the PGA chips. The Digital ground for the digital parts of the circuit, the center tap of the power transformer, and
the "shield" circuit, which protected the signals but has no current path otherwise.
All 4 of these grounds were joined together in one place only, the center of the regulator board.
In this design, the EQ is purely analog part so I'd think a single AGND is sufficient. Furthermore, the 'ground shields' leading between audio signals are not leading anywhere so they might really be considered 'shieding traces' as mentioned above.
I also used separate regulators for the analog +5 and digital +5 circuits. to reduce digital noise.
I have these, too. A weirdness happens when I connect grounds of the two power supplies (+/-5V analog and +5V digital) - the analog PS starts drawing as much as 100mA (@230V), but luckily I have lab PS to test with. Power supplies will be designed in the end when I know all the currents needed and all the place available.
I'm diving back into Java this morning to try and get the mix window ready for release.
Once the hardware prototype is ready I'll start programming the DigiPot (AD5206 in my case) part, rest is done already including bus switching. I have also modified the Java for testing, I'll send it back to you if you want it, but as discussed elsewhere, there were just minor modifications - extending channels and adding few more controls...
I know the PCB still is not perfect, but a little go/no-go might help :rolleyes:
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P.S. Of course, once the prototype is built I'd be happy to share the schematics and source code!
So here are all the schematics and PCB designs, I was busy designing and creating the PCBs during Xmas (I luv Xmas :rolleyes:).
However, after building the EQ prototype I found out that it's much too noisy, even with no pots connected I'm unable to get square output from square input... The best result is attached down there. Apparently the PCB design causes the noise (however there was no noise when the EQ was built on a vector board), are there any audio PCB design rules I should be aware of? I'm planning to use shielded cables for connecting the pots, I can also try leading a grounded isolated wire between each two audio signals for test (input, output and pot connections) next to analog ground? I can try to modify the already-built PCB but would like to know whether it actually helps before I re-design (and etch) the PCB...
Below is the EQ PCB design as well for quick reference.
Thanks.
P.S. If anyone is thinking about building such mixer I'd encourage to use AD5204 instead of AD5206, though more ICs will be used AD5206 has no SDO signal and doesn't allow chaning, that's the reason for all the /CS logic found on 2nd sheet... I do already have 20x AD5206 so this is no choice for me :wacko:
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Very nice, Good Luck!
I hope we get all the details as this unfolds.
It's going to take a lot of knobs to get control of this. If you'd like a tool to help out before the MB64 is built, I have java code here for a mixer window.. It would take about three minutes to add six EQ knobs to each channel.. I'd be happy to send you the java code, or even to add the EQ knobs first, if you'll tell me what Control Change numbers you're using for each band.
In any case, it looks great.. I'm envious of your project!
Lyle,
thanks, though I have Korg 49 for MIDI control, the tool you're mentioning would definitely help. My current MBMix CC assignment uses CC#91-CC#93 (for old-to-be LPF, BPF and HPF), but since there will be six levels to control plus per-channel Volume (CC#7), Pan (CC#10) and four global L/R buses (not decided on MIDI implementation), I'll use Sound Controller 1-6 (CC#70-CC#75) for controlling the EQ. I'd be happy to use the tool, I didn't figure yet whether the Korg 49 is able to use MIDI channel 'shift' for 8 encoder/8 fader controllers.
Looking forward, thanks!
P.S. Of course, once the prototype is built I'd be happy to share the schematics and source code!
P.P.S. The project wouldn't even start if you didn't start fiddling with PGA, not even talking about TK and the whole MIDIBox :rolleyes:
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Hi,
after several unsuccessful tests with different filters and crossovers I've decided to go for basic 6-band equalizer. For setting the per-band level I'll use digital potentiometers. Since there are 6 bands I'm going for Analog Devices AD5206. The equalizer is a standard one with op-amp input and output buffers and gyrators to substitute inductors in LC pass-band filters. The component values to be used were computed based on my in-shelf supplies to re-use maximum amount of filter capacitors bought for different filter and crossover attempts, but basically the bands are 50Hz, 100Hz, 200Hz, 800Hz, 3200Hz and 12800Hz. Resulting audio signal will be connected to two PGA channels (left/right) to control channel volume. The mixer will be controlled via MIDIBox 64 (or 64E, though 128/128E would be useful for controlling 8 parameters for 16 channels :rolleyes:).
THe AD5206 uses the same SPI control bus as PGA4311 is using, so I'll start making the modifications to the software as soon as the equalizers and PGA channels are build. There will be 16x AD5206 and 8x PGA needed for 16 stereo channels plus additional 2 PGAs for total 4 stereo output buses (as mentioned in the previous posts), so the SPI bus control signals (CLK and /CS) will definitely need to be buffered.
For results see the thumbnails below, they are (in the following order):
- response for 100Hz sine wave, the original (unmodified) wave is cut down in half to see both waves
- response for 100Hz triangle wave with original cut down
- response for 100Hz pulse wave with original cut down
- response for 100Hz "equalized" pulse wave with original cut down
- response for 1kHz pulse wave with original cut down
- response for 10kHz pulse wave with original cut down
- my current desk top with the EQ circuit built on vectorboard (no PGAs and no AD5206s yet)
A slight distortion can be seen on all waves (because of shutter speed you may notice the higher-amplitude signal being "thicker" than the other one), it can be seen best on the 1kHz pulse wave. According to the tests performed this is caused by the wires connecting potentiometers with rest of the circuit, there is no ground on the wire. When AD5206 will be used, I'll take care and make the audio lines as short as possible, when possible with analog ground surrounding the lines.
Wish me luck...
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got it, thanks!
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Strophlex,
now I've got the impression that I'm lost in your answer ;-)
So these signals have a frequency of 10kH and 100kHz and where sent through a hpf with cutoff frequency of 2kH? Looking at the square waves, I think they look low pass filtered, the 10kHz one more than the 100 kHz one. Is there some low pass stage in the filter you sent them through? In that case the 100kHz signal should be more affected then the 10kHz so it doesn't make sense. It's still early here so my head is not started yet. Anyway, I think the second osc. probe will show something.
EDIT: I looked at the wrong pictures... Both square waves look hp-filterd. Looks like a hp-filter with cutoff frequency at well above 100kHz. Did you look at the signals through the amp only yet? I don't understand this
Not sure what you mean by 'amp' (maybe the op amp?), the only 'amp' that might possible involved in the setup is the PGA chip itself. The current setup is either:
(final setup)
function generator -+- LPF -> PGA channel -+- oscilloscope +- BPF -> PGA channel -+ +- HPF -> PGA channel -+or (filter testing setup)function generator -> LPF -> oscilloscope -> BPF -> oscilloscope -> HPF -> oscilloscopeThe waves above were measured with this testing setup:function generator -+- LPF +- BPF +- HPF -> oscilloscopeSince the waves were 10kHz, resp. 100kHz, besides the "substraction" filter construction, I'd assume that the signal passes only through "HPF" though "HPF" is composed of two "time delay" units and summing unit (confirmed by the oscilloscope - no signal was on BPF or LPF).
Hmm, looking at the schematics (here, for quick reference), the BPF takes input signal, filters high frequencies using 3kHz LPF and substracts the already filtered LPF signal, resulting in band-pass filter 200Hz-3kHz. Following the same logic, I'd assume the HPF should take delayed input signal (connected to positive input of IC7/B) and substracts BPF signal (connected to negative signal of IC7/B). The delayed signal connects to negative input, however. Am I missing something? If there is no reply for this during the day, I'll try to connect it (as it makes sense to me) in the evening... Strange thing for me is, that the filter does actually work, for sine wave, that is.
"Since they're for some reason distorted already at filter output it's clear that PGA can't do anything about it... "
Sounds like you assume the hp-filter works. Why not just connect the scope to the output of the amp if you didn't do that allready.
Again, not sure what you consider being an 'amp' - op amp? The osci is connected either directly to filter outputs (i.e. op amp outputs plus some resistors/capacitor) or after all PGA channels, which has all the filter outputs mixed. The snapshots in my previous post were done with osci connected directly to filter output.
Thanks!
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Strophlex,
It makes sense that the signal looks distorted. A sine wave will probably be passed through one filter only and thus there is no risk for interference due to phase differences. The other signals are composed of many frequencies. To get a complete square wave out you'll need to have the output of all filters added together, in phase. Thus, if you take the output of one filter only, it will not resemble a square wave, especially not the HP-filter. Attached a matlab plot of a hp-filtered square wave. Excuse me if this is old news...
the reason why a filtered non-trivial wave (triangle/square) is distorted is clear to me, the formulas can be found on Wiki or in my school notes. This is not the case, unfortunately. Well, it is, but this is obvious.
The problem in my case is that (after new measurements were done) non-filtered complex wave is distorted. In the attachment below, there are two 10kHz waves passing through filter and measured at filter output. The HPF crossover point is 2kHz, so the waves are far above it (10kHz), therefore passing only through HPF. Since they're for some reason distorted already at filter output it's clear that PGA can't do anything about it... I only have one osci probe so I cannot make ALT snapshot of the two waves, but I'll get second tomorrow, promised! :rolleyes:
Maybe this is given by the filter construction - I don't know the english word but the filter itself consists of two LPF filters (and time delay units) and the filtered signal is substracted from the original signal in subsequent stages, so since there is phase difference (if phase difference is what I see on the osci) - this is the only place distortion can occur. This filter was chosen since it has 'linear phase', but apparently 'linear' doesn't mean 'flat'... Apparently, I need to make more filter experiments, anything to suggest? Bessel filters seem to have flat phase delay, they'll be probably my next choice for testing...
- check and measure the power source during triangle and pulse
- check and measure the clock
What did you mean by these? Direct generator measurements? These are as straight as expected with no distortion, so the generator works as a charm. Or?
Thanks for more info and ideas!
What is interesting to me is that for 100kHz the waves are much more what they should be. Is there an explanation for this? (See latter two thumbnails)
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Did you use the component values sugested in the article? Did you look at the different filter outputs paralell on different oscilloscope channels? I think it looks like the channels have different phase after all, but I am absolutely not sure about it.
Yes, same values were used, even the 56.3k resistor is 56.3k, rest is within tolerance of the components, though I know that audio applications are quite sensitive to tolerances, but since MKT capacitors are +/-5% at best, there is not much to choose from...
Hi,
I am also not a DSP specialist.
My suggestions:
- set the oszi to AC and measure again
- check and measure the power source during triangle and pulse
- check and measure the clock
Hmm, the osci is set to DC (I didn't see any difference for sine wave, that's why), I'll change it to AC. I didn't catch the difference, anyway :rolleyes: . I'll take a test look in the evening, thanks for hints so far :rolleyes:
Do you use a amplifier?
If you consider PGA being an amplifier, then yes, but the freq response should be okay for PGA, according to datasheet and web page it's used in professional audio application. Anyway, the interconnection looks like: Functional generator -> filter -> PGA -> Oscilloscope.
I never checked whether the output of filter directly is distorted for non-sine signals or not (I doubt it'll be distorted), I always connected the osci after the PGA amplifier. I'll do that when I get back home.
I did more checks meanwhile and found out, that when a signal passes through a single filter (>10kHz, i.e. only HPF is involved), the signal is distorted as well (measured at PGA output as mentioned above), do the phase and delays are not involved in this case. I'll see tonight whether the filter output itself is distorted or whether PGA adds the distortion...
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I think your best bet is to write an eMail to Greg.
The only post I found was performed as a guest and mail from his email address mentioned in MIDIBox gallery was rejected. So, again, does anyone happen to have the files?
Thanks
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Hello,
I'd like to build 64-pot Midibox64, but since I'm not a good PCB designer (as mentioned elsewhere) but I'd like to have a single PCB I'm looking for several times mentioned Greg McMillan's MB64 PCB layout but can't seem to find it. I know it was designed for pre-MIOS hardware but since PCB erratas do exist I'd like to re-use it, anyway. Does anyone happen to have the files?
Thanks,
freddy
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I used "Schmartboards" successfully to prototype the circuit. They worked very well.
My desire for less stages is about sound quality more than core count.. I don't know of any quality problems with
multiple stages, but I have never tried, either.
As mentioned before, I'm using ZIF socket for SOIC (from my EPROM programmer). I only have a single one and it's quite expensive (60EUR), I doubt I'll buy more. But since I definitely know I'd like to build PGA-based MIDI mixer, as soon as I make a PCB and solder the PGA (I'll do that at least for a few channels regardless whether I'll use single- or double-stage) I'll free the ZIF socket and will continue with multiple-stage experiments.
The software is a bit over-commented.. some find that annoying, but it is the lesser of two evils.
Put simply, incoming MIDI messages that affect the mix are stored in a big array, and any channels that may
change as a result are flagged for re-calculation.
Then the "math" part processes each flagged channel, combining volume, expression, master volume, pan/balance,
effects levels, and effects flags into a single gain setting for each PGA gate. Finally, after all calculations are current
with the MIDI data table, the gain settings are written out as a big block to the PGA chips.
Of course there are other minor routines.. save and restore bank settings, display channel names and gain settings,
responding to MIDI requests for the current state of the MIDI table.
After a few tries I managed to get the functionality as desired, thanks for explanations and to source comments. There are still few flaws in the implementation, but since I'm using a single channel for now it's usable, though.
...
if you went to a 4 band EQ, there would be one PGA chip for each audio channel,
much easier to divide up as you build.. (though I think the software could handle it either way)
I have written enough for now.. I'd love to see some discussion on this when I get back from work tonight.
Up till now I as experimenting with different filters. First, I used a simple 3-channel Butterworth filter (http://users.otenet.gr/~athsam/3_way_active_loudspeaker_eng.htm) but this one suffered from phase shifts and time delays. The resulting sound distortion was audible even by me, this is a 6dB/oct 1st order filter with no linear phase, so this is obvious. Too bad for me that I already bought components for several channels. I'll try to reuse them. Hmm, it's been quite a time I left the uni and I wonder where my calculator is?!
Currently I'm using 3-channel linear phase response crossover filter (http://users.otenet.gr/~athsam/3way_active_crossover_with_linear_phase_eng.htm, really a great spotted site, Lyle!) built on a vectorboard:
(I managed to fry few of the chips, does anyone know if TI keeps records of the sample they sent? Can I ask for more samples? :rolleyes:)
For simple sinewave the crossover works as a charm for the entire 20-20.000Hz spectrum, but for more sophisticated waves (triangle and pulse) the resulting output is distorted:
(1kHz triangle)
(1kHz pulse)
The waves remain far from their meant-to-be shape regardless of filter trim-tuning... Hopefuly someone with audio (and mathematics) skills can tell me what harmonics are missing or are much too loud...
However, for really complex waves (music) the distortion is barely audible (at least by my ears and more-than-average quality headphones).
One more thing to ask for, if the troubles are solved and I'd be thinking about building the PCbs for multiple channels, I wonder if anyone could help me with PCB routing? Of course, I'd give out the schematics in Eagle but I'm really unskilled with PCB design and effective components layout. I'd really be thankful if anyone could give a hand...
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Lyle (et al.),
I'd like to build 16 chanel full audio mixer with no effects but with 3- or 4-band EQ on each channel. I'd like to place some 3/4-band filter before each channel board to allow control of per-band volume. The idea is that EQ tone characteristics would be controlled by 3 or 4 different CCs, the resulting 16 channels would mix into resulting sound controlled by volume/pan CCs for the same 16 channels.
For the input channels, I need 16 boards (possibly 2 cores with 8 channel board each), for post-input 16 "equalised" channels another 4 channel boards (with 3rd core) will be needed. Since there are still just 16 audio channels altogether, standard MIDI channel to audio channel mapping would be used (though it's no problem to workaround this limitation as stated by you several times). So far, so good.
What is starting to confuse me are the audio busses to connect to. After reading channel board info, your recommendations regarding line vs. full audio mixer it seems to me that buses will be not needed between the two sets (per-band input and post-input) boards at all, but all PGA outputs for a single channel (i.e. for per-band input channel board) will be connected together forming a single audio feed for second set of input boards (post-input). These post-input channel boards will control volume and pan of the resulting sound and therefor will have L/R busses which will be mixer output connections.
Did I get the channel/bus idea right? Is this concept practical or did I miss something while thinking about it?
Thanks.
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Got my batch today, let's get back to soldering ;) Thanks TK!
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Folks,
I have two different C64 PSUs, one is from C64 (whose box is hosting 4x v1 SIDs) and a C64-II PSU. When powering either 4x v1 SIDs or 4x core v2 MB6581 with the old C64 PSU, everything works as a charm.
When using C64-II PSU, for the 4x v1 SIDs I get _lots_ of humming in the resulting mixed channel (but the SID modules do work, actually), v2 MB6582 even doesn't power up properly - the 4x20 backlit LCD gets lit, first and third line are blacked out and I cannot get any further, no CS LEDs lit, but the humming can be heard.
When I measure the voltages, +12V for SIDs is okay in both cases, but instead of 4.99V I'm getting with old C64 PSU I only get +4.45V for PICs, 5V fan, HC595, LC256, etc. When I measure unloaded PSUs, for the C64 (old one) I'm getting +5.22V, for the newer C64-II barely +4.03V.
I suppose this humming is caused by low voltage, as is the "unability to boot" (because of higher load of v2 MB6581), but can someone audio-skilled confirm this before I start opening the C64 PSU? Any similiar experience (i.e. to what to look at inside the C64 PSU)?
Thanks.
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No, this cannot be "fixed". Displaying realtime informations from the slaves would produce too much CAN bus traffic.
Sure, makes sense. Thanks!
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Not sure about the first problem, but number 2 is just how it works. Leds only trail on the first core.
Can this be "fixed" (i.e. extended to other cores as well)? I can not imagine a reason why it shouldn't behave the same for all cores.
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So far no problems with MBNet/CAN. Thanks for pointers!
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correction: though everything seemed to be in place as stated in my previous reply, re-flashing SID firmware to the SID3 core seemed to help at least with the MBNet. I'll wair for the CANbus errors to occur again, if ever.
Thanks.
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Thanks for reply, but:
- R80 is in place
- none of the jumpers seem to be missing (which one did you mean, in particular)
- firmware is in there - connecting the LCD to each of the cores (save the master) yields the same message - banner message and info about not launching CS.
Sub-millisecond wait routine
in MIOS programming (Assembler)
Posted · Edited by freddy
nILS,
by lacking free pins I need to employ DOUT to interface digipot chips using SPI interface, this makes 2 wires + /CS signal. The code doesn't really work as expected so I need to hook my logic analyzer to the bus. The "logic analyzer" is built from LPT port (I had trouble to find it these days :rolleyes:) and therefore .1us (or similiar) delays are not really seeable by the LPT (I only got some 800kHz sampling rate).
I'm using MIOS_DOUT_PinSet0/1 routines to set the SRIO value (so it takes definitely more time than .1us) but I still can't see the toggling signal, so I need to insert some delays in order to be able to monitor the SPI and debug further. But inserting MIOS_Delay of 1ms is too much since there are tons of bits and then the MIOS is really slow and busy spending time in delay loops. That's the point, once the code is tuned and verified I'll probably abandon the delays if the digipot chips are able to react (the CLK rate must be at least 20ns according to the datasheet so it should be no problem).
Does the 1ms SRIO mode refresh time mean that I can't really toggle DOUT pins faster than in 1ms interval?!