jrp
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Posts posted by jrp
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Hello, yes! It feels very nice now! Thank you for doing this, now a lot of people will have even more fun with their sid!
Seems like i should get different Encoders though. They do work, but when i turn slowly i get solid one step increments for a few clicks - and then a sudden jump of about 20 steps. There is no difference when using the main encoder.
But besides from that, tweaking really has a better feel now. :smile:
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edit june 6 - more sound examples
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The Midibox-Project has been an inspiration to me for quite some time. While the most simple controller is possible, i normally cannot resist going big...
Over the last years i have mainly been designing compressors and preamps for studio use.
Before that i was working on a modular synth. That´s what i started with, that´s how i learned electronics. Mostly thanks to electro-music.com, music from outer space and some other places.
CMOS was the only thing digital i could work with untill i discovered ucapps.
This post is to discuss and present my take on an analog signal chain for the Midibox Sid.
I composed this like i always design - using building blocks from datasheets, standard opamp circuitry, "known" designs, blocks borrowed from other people, well, sometimes i even invent something (sortof).
It features
- CEM3379 Filters (but these circuits could be implemented with any vcf)
- voltage controlled Filter FM:
fm sequence-.mp3
fm 1-.mp3 These two clips use Filter FM controlled by LFO, tweaking the pitch of the modulating OSC.
- voltage controlled Overdrive with volume compensation (needs improvement!):
drive.mp3
- a wavefolder, designed by Ken Stone with some minor adaptions. http://www.cgs.synth.net/modules/cgs52_folder.html
waveshaper.mp3 there is a short bit of an clean sound and a bassline, then they are both shaped. At the end the bassline is doubled by the same patch, 1 octave up and with a saw instead of tri.
folder single triangle lfo fade.mp3 This clip fades in an LFO controlling the folding of a single triangle osc. The folder is turned on after 4 bars
folder 3oscs lfo mod.mp3 Here three detuned OSCs are used. OSC3 is a saw one octave above the other two tri OSCs. Sustain was turned down a bit to get a good responce from the Folder. Again first 4 bars are clean.
fm fade and folder.mp3 Single Tri OSC playing. Right side is tuned up 2 octaves and used as FM mod source. FM is controlled by envelope, depth is faded in over the first 12 bars. Then the folder controlled by an LFO on the other cv channel is switched in
- fading from LP to HP of external filter
Of course Sid already has a HP Filter mode, but this was just so easy to add... see schematic below.
HP SAW 0-12-24.mp3 Saw OSCs pitched 0, -12, -24. High cutoff, HP is slowly faded in
HP pulse +12-12-24.mp3 Same, but Pulse with PWM tuned +12, -12, -24
- external ins
examples are a bit noisy couse i haven´t build the line reciever yet, so it´s just a signal hooked in with alligator clips...
ext pad mod filter vca fm.mp3 A Drumloop is tweaked with the SID engine. OSC FM is used in the second half. Note that there is some bleedthrough of the modulation osc.
ext pad mod filter vca fm.mp3 This time a pad sound was used.
- additional noise + vca
- Main VCAs, mixer and balanced line drivers
Here are some basic waveforms. You hear three short clips with modulated wavefolding. Then there is a short sweep of distortion and then a couple of FM waveforms. All were generated using a single triangle oscillator.
--sorry, file was to big, have to bounce it again----
Although this is still a work in progress i already have a bunch of schematics with tested values so i thought it´s about time to share a little.
Soon after i hooked up my first Sid-Chip to a core module i was hooked up myself. I spend a lot of sparetime over the last 6 month, thinking what could be possible, what could be usefull, what could be fitted into the MB_SID concept.
The analog CV-options and triggers are perfect. Thanks TK.
I use analog filters and VCAs, they eat up 6 CV channels.
Leaving 2 CVs and 8 triggers.
Not much if you are used to modular synths. But that was part of the job- selecting the right blocks to get the most out of what is available. And i soon realiced that you would rarely need any more.
Oh - and everything is saved with the patch!!!!!!
I tried a bunch of circuits and ideas and came up with this collection of building blocks.
Among the design goals were:
Usefull addition soundwise (very subjective)
Low parts count (more or less)
Effective controll with one CV per stero side.
All CVs are 0-5v, the DAC can drive 2k load so i dodn´t use any buffers. Of course a complete AOut Module can be used if you like to try any of these circuits, but if it is calibrated to 10,xx volts the cv inputs should be adapted. If you know what a voltage divider is this should be no problem. I use trimmers anyhow, but with series resistance to taylor the trimmer to a more precise action. Lowering or raising the coresponding resistor does the trick.
Effects can be switched in individually, but share the same CV so the same modulation will be applied. Can sound very cool, but normally one at a time is enough for a certain sound.
There is also a mode that controlls two sets of effects - in mono.
Have i said i love the modulation matrix?
Sorry, i have schematics, but no board layout. I am a veroboard guy... I refuse to learn any layout software...
I am happy i found a good way to solder SMD. I carefully bend every second leg upwards. The remaining pins can be soldered nicely to a veroboard. The bend pins will be soldered using short bridges (cut resistor legs). I use this a lot to perform crossings (like multilayer) or connect ic pins, etc. Quite sturdy, of course you musn´t apply preasure, so whatch out when changing chips. It´s definitly quick compared to strapping wire.
Anyways, i like working like this when there is a good audio book around. Kind of fun looking for the shortest signal path as you go along filling the board
.
In the current state i have implemented and build (in stereo):
- CGS Simple Wavefolder with some adaptions
- Filter FM
- Highpass blend for external LP-Filters using same VCA as Filter FM
- Voltage controlled Overdrive with volume compensation. Needs improvement.
- Switching of modulation Signal for FM sounds. Making Sid L the source for both signal channels, Sid R the modulation source for Filter FM.
- Stereo/dual mono CV mode
Only checked out, under construction:
- Noise for external Filter
Todo:
- Tweak the overdrive for less noise and better Bass-responce
- build signal switching circuitry
- design frontpannel and CS
- decide if F2A should be used. This is a problem in conjunction with the 8580 requiring calibrated log mode. Alternative would be to include independent controll for CV 1 and 3
- put everything in a box with psus
- decide what could be ommitted
- Build Stereo Sid 2,3,4....
Here you see a block diagramm of the signal flow.
-A buffering amplifier brings the sid output to a more roboust level.
-Folder, Filter and Overdrive are chained and can be switched in and out (see extra diagram below).
- FM and Highpass are part of the Filter-schematic
Signal and CV switching
Most of this is propably self explaining. Switches controlled by the Gates from MB_SID. Stored with the patch. Very nice!
Switches are not selected yet, but i´ll propably go CMOS 4066 or 4053 for CVs and some Maxim Chip for Audio-Signals. Max 333 ACPP might be the choice. For 5.20 € at reichelt you get 4 SPDT switches that can be powered with up to 30v. For a compressor i would always use a relay, but this is a synth...
Maybe powering the 4000 series cmos chips from +-7,5v would be smart. makes the PSU more complex though.
I still have to test how they perform with AC-coupled signals when powered from 12v single rail.
SW 2 is different for L and R channel.
This will route the left Sid to both audio channels, making the right Sid the MOD source for FIlter FM of both channels.
SW 1,4,5 are controlled by a center off, DPDT switch on the panel.
They provide three modes for external signals:
EXT to SID_input
EXT to CEM3379 filter directly
EXT to CEM3379, Sid disconnected from filter (but still available for FM!)
To be honest, this should work, but i might not build it. If i really build 4 stereo voices, they could be fitted differently.
SW 11,12, 13 are also hardware controlled for routing Voices 2-4 to the Main out.
A Gate is used to split CV 7 and 8 between L and R channel (each cv controlling every active effect) - or - direct CV 7 to Wavefolder and distortion, CV 8 to the other effects - on both channels.
Wavefolder:
I found this little circuit on CGS. Really fascinating what it can do.
http://www.cgs.synth.net/modules/cgs52_folder.html
I found that, although this will react very much on different input levels, it sounds best when a defined level is used (so no VCA is required). I calibrated mine to work perfect with a single oscillator. Using multiple OSCs sounds interesting as well, but some sounds are "broken". Luckilly oscillators can be turned down with the Sustain.
Instead CV is applied as an offset. On the schematic i tried a drawing of the waveform at different CV levels.
It sounds a bit like a phaser and a bit like pwm for triangle and saw.
Filter, FM and Highpass:
Depending on Switch SW 2 L, R a modulation signal is routed to an inverter and VCA. FM is generated by switching the VCA output to the Filter Cutoff pin. HP by summing the VCA output with the filter Output. Since the VCA delivers an inverted signal it will cancle out below Cutoff - creating a fade to a more or less Highpass responce. Note that, if switch SW2 is on, or the wavefolder is active, things look a little different.
The rest of the Filter is more or less off the CEM3378/3379 datasheet. I settled for a slightly higher drive, giving this filter a little more grit.
Of course the same circuitry could be added to any VCF.
I have not yet implemented the Noise. I tried it once and liked having white noise as an alternative to the sid noise sound. Also having an extra VCA for noise is nice to have. Schematics for analog white noise (using a transistor with open collector) are on the web.
Overdrive
This can be seen as first design concept. I have to do something about the noise, propably by chaining more clipping diodes, thereby raising overall level.
It could also use some basic filtering at the output.
The overdrive itself is made by the opamp with the diodes in the feedback path - forming an unsymetrical sof clipper. You know this from the guitar stompboxes...
The first VCA acts as a voltage controlled drive-pot. The trimmer around the CV opamp will set a bias at low level so with no CV applied there will be close to none distortion.
Volume compensation is archieved by applying a portion of the CV controlling the Drive (first VCA) to a second SSM2164 VCA but without the inverting opamp, turning it down as volume raises. Remember SSM2164 is at unity gain at 0v cv and will turn down with a CV going positive.
This works very well, it can be trimmed to taste so that the percived Volume is constant or if you like, slightly raising.
If i had a 9th gate i would make it switchable pre or post filter.
But that could also be solved by building different Voices.
Pre filter is nice as you can controll the harmonics, Post filter has a more agressive touch and gives cool clipping of filter selfoscillation.
VCA and Mix
Nothing special here, only maybe that i use an unbuffered CV that is inverted in the ensamble settings. Nice feature!
Since my Studio is all balanced this is what i added for outputs.
compensation caps (47pf) across the output opamps feedback resistor are required and missing in the schematic. I use NE5532 for it´s 600 R drive capability.
Sorry if this was confusing, i am a bit hit by how much info this is myself... -
i just did a quick test for now as i am off for work. I am not quite sure if i notice a difference. Was it a big change on your setup or rather subtile? My test is trying to set cutoff to 800, i most often get jumps like 76f to 830 with just 3 ticks of the encoder unless i turn very slowly.
Do i need to upload the new mios to all slave cores as well?
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wow! you are incredible.
I read your first post and was already forming thoughts like "oh, what a pitty, i hoped it was a simple adjustment". No surprise, i cannot "read" assembler but i have read about it....
And then that second post :smile:
So you changed it by altering mios8 if i understand correctly. Will this have an effect (positive or negative) on other applications, like MB64e?
I´ll try this as soon as possible!
Regarding Steps: I recorded some sounds today and was tweaking the synth, quickly editing. Fast mode feels a lot better to me now. Some of the stepiness i heard was exagerated because i was controlling external vca that was poorly calibrated and was missing a smoothing cap at the controll pin.
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the demo makes me think i should give feedback another try.... like the scream it gives.
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If you start experimenting it is important to use a cap to ac-couple the signal to the input. SID input pin is floating at half supply voltage, not ground.
i am designing the rest of my analog signal path right now, so this has to wait a little, but i am interested in trying.
Basically it allows you to process analog signal like drumloops with all the goodness the SID (or external voltage controlled effects) have to offer. So it´s really cool! I already implemented this with my external filters. You can imagine what you can do with a drumloop, tempo synced LFOs and envelopes, using filters, distortion and vcas....
I expect the sound of the sid input to be a bit grainy compared to using an external filter. Maybe it has a cool vibe to it when going beyound the 2vpp signal level pointed as maximum in the datasheet. And the sid filter of course...
I definitly want to check it out, but as i said, for now i am at the other end of the signal chain so to speak.
About feedback: With my 6581 this does very little but adds a bit of grain, distortion and boost you might find pleasant. 8085s will go into oscillation when in BP mode.
I decidet not to implement it as i am using external filters and a lot of much more powerful voltage controlled processing so it had no reall benefit to me.
If i would implement it it could be considered to use a fixed feedback level. At least for me i had the feeling this was more of a turn on turn off type of effect. - In other words, my pot during experimenting was either fully up or down when playing with this.
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Yesterday night i had the time to play with my MB_SID and get a better impression of the new feature.
The Speed is actually good - turning an encoder two times for the whole range still feels very resposive. In audio ranges are very big...
The "steppiness" does not feel so good, unfortunately, especially when trying to do small changes.
It seems to me that the transission between slow and accelerated speed is quite low. When an encoder is turned slowly the increments are smooth and slow. Go just a little faster (but still quite slowly) the threshold is reached and speed is accelerated. If this threshold would come in a bit later that would feel much better.
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We shoud stay on topic, you are right.
About different speeds for controllers controlling the same range - this is not really necessary. But it could come in handy to really tweak the responce of the CS - i cannot really tell by now.
I am also really interested how other users feel about this topic, and what types of encoders they are using. I was searching the forum a lot before because i thought i simply had bought the wrong encoders.
Thanks for pointing me to the right direction. I can very much understand your point not providing this as a general configuration option. I find it also very positive if some hints are given where to look instead. Might be a lot of trial and error for someone like me, but it gives the interested a very little gasp on how a sourcecode like this is functioning.
In CS_Menu.inc i found this, i figure this is the place where the responce could be adapted for different types of encoders:
CS_MENU_EncSpeedSet_ModVal_GT1F
;; max value > 0x1f: set fast speed depending on range:
movf CS_MENU_PARAMETER_MAX_H, W
andlw 0xf3
bnz CS_MENU_EncSpeedSet_ModVal_GE400
movf CS_MENU_PARAMETER_MAX_H, W
bnz CS_MENU_EncSpeedSet_ModVal_GE100
movf CS_MENU_PARAMETER_MAX_L, W
andlw 0x80
bnz CS_MENU_EncSpeedSet_ModVal_GE80
movf CS_MENU_PARAMETER_MAX_L, W
andlw 0xc0
bnz CS_MENU_EncSpeedSet_ModVal_GE40 -
i could only do a quick test right now and i can say cutoff and depth work much better now! This is really cool! Could still be a little faster for very fast wide speeps, but on the other hand that would be best done by a pot anyhow... so i think your efford on this toppic was a great success.
Is there a way for the user to configure the reponces further within the firmware to adapt for different encoders or is this simply to difficult?
I have the impression that envelope2 is messed up/not working on some of my patches, but i assume this is due to some error on my side again. These patches had been made with a modded firmware where i had dec 1 and dec 2 flipped in the menu for conventional ADSR responce.
Unfortunately i´ll propably not have any time over the weekend for further testing. But for now, it feels really good tweaking the knobs. Envelope is working when setting up a new patch.
Regarding envelopes (please just say to complex - i don´t want to steal your time - and i wouldn´t want to start a new thread asking for stuff): I have been searching the code with no success for a way to double the envelope1 depth within the modmatrix. Right now i assign env1 to source 1 and 2, using 1+2 opperation like described in the manual, to get a full-range cv sweep to my vcas. So to add more amplitude modulation (eg multiply env with velocity or a lfo) an additional modulation path has to be used.
Also i was trying desperate to change modulation target osc pitch 1,2,3 to ext 7,8,Q simply couse i use these more and hate not using half the cool modmatrix.
Not important, it works nice this way, but i just can´t resist trying to tailor your application to my setup as much as possible. Call it an obsession ;-)
No idea if these are easy tweaks or require larger code rewrite. If i get a hint how to figure it out i would be glad to write a easy to follow step by step guide for the wiki. I see these things will make the application incompatible to existing patches and are only for a user who can live with that.
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I just played with this a bit more. My setup is including CEM3379 filters.
Sweep could even be a bit faster than speed 7 but it´s nice like this already. Unfortunately tweaking the knob does not produce a smoth filter sweep. It sounds more like a glissando (=stepped).
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The problem i see is that with different encoders the responce will be different. Combining SID with MB_NG sounds very exciting indeed.
About my SID not outputting sound with the new firmware: I simply forgot that my ensamble settings were overwritten by the upload (for now i have only one bankstick - i´m prototyping analog enhancements at the moment). I was simply sending on the wrong channel ;-)
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Sorry, i try to stay on topic, but this is just too exciting...:
In addition, I will add a "CC dump request" which will allow to synchronize a MBNG with the current patch CCs.This sounds very interesting. So it could bring ledrings to our sid? Among other things... Analog pots in snap mode maybe...
But isn´t a CC always a 7bit value?
I feel especially cutoff modulation will suffer from lower resolution. The range of sound-frequency is simply enormous.
In the sysex implemention i read this: Is it not the same? I was scratching my head about the possiblilities for a CS in combination with MB_NG
01/b) F0 00 00 7E 4B <device-number> 01 08 00 00 F7 Request the current patch edit buffer (direct read from RAM)
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Hello,
Thanks for putting this up!
The testmode works perfectly but i am missing sound from my sid after uploading. Is that intended? I had to test the speed modes relaying on display rather than sound feedback. This makes me a little unsure.
Just for my understanding: The firmware will read how fast i turn the encoder and act accordingly, so there will still be full resolution, no audible steps, right?
The firmware seems to switch between two speeds as soon as a certain turning-speed-threshold is reached. Is this correct?
Here are my findings with 20 ppt detented encoders from unknown manufactor (i assume these values will differ if other encoders are used):
First of all: I am very happy!!!
For Fast mode:
- Cutoff:
Speed 7 - it still takes 2 full turns to cycle through the whole range. So it could even be a bit faster. But since i couldn´t hear anything i don´t know if it will feel smoth. But i know it is actually handy to dial through the whole range with the twist of a finger sometimes.
If i consider an additional analog pot for tweaking (from the knob function or from outside via NRPN) than 7 is propably perfect for normal editing.
- Resonance:
Speed 4 - 360 deg turn. Could be a little bit faster, but seems best value for me.
Speed 5 - a bit faster feel than a analog pot would give. That makes it too fast definitly.
- Depth:
Is actually the same range as resonance, right?
I would use speed 4. Maybe 3 is better since it is split around 0 so it will normally require less fast tweaking. Really would like to hear how it reacts.
4bit Sustain:
Speed 1 is nice, perfect would be inbetween 1 and 2. Actually normal is also fine for this :)
Slow Mode:
I personally propably would not use it as the ecoders already sense the acceleration. Turning the pot slowly works well.
Bisides that, for all the 10bit, 8bit and 4bit parameters Slow 0 or Normal (is it the same?) seemed useful.
Is there any chance these values will be adjustable by the user in a final release?
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Hello Thorsten,
Thats very nice! I would be glad to do some testing! SID is awesome!
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changing the code is really simple. Just change the line and recompile. I already tried it but for me it changed nothing. So it would be really interesting if it´s a mistake on my side or if it simply doesn´t work like that. Maybe just a little mistake in the discription (although i would normally trust tk blindly).
So people, if you are interested, just try it and tell us how it went. If it works for you i´ll have to keep trying and see what my mistake was.
Besides, i still belive using a MB_NG or MB_64e to handle the controllers would give more options and would be much easier to tweak for best feel and resolution.
The only thing that troubles me is that a second core running the encoders or pots would have to determine what SID (1,2,3,4) is currently active so data will be transmitted on the right channel. Luckily each SID engine sends a dout flag when active, that´s where the SID LEDs are connected. This could be used to let MB_NG switch to the right midi channel(s).
This would also mean that channel assignment would have to be fixed in the ensamble settings. So a Midi router would be required since we want to be able to play each sid individually or unisono from one channel.
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possible we are just using the wrong encoders?
I built my cs prototype with cheap encoders i ordered from china (about 15 pices for 5$...). They are detented (24 clicks) and detents cannot be removed as the detents are made of the switching contacts.
Somewhere i have read that by using nondetented encoders the resolution is much better, although i don´t see how this will work as they should still only have 24 impulses per turn (right?).
Maybe someone has some input on this and can recomend a better suited part.
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I have been working on some minor code changes (mostly regarding new buttons and encoders) and can tell that it´s not easy. Assembler code is easy to mess up i guess... But i am no programmer, so maybe for you this is more straight forward.
Anyway, if there are alternative solutions that´s fine with me. And NRPN offers full reolution for all parameters so it´s the best choice anyhow.
I see a good idea behind the cc handling for several situations:
If you use a modwheel, do you want it to change your patch?
Or if you save after playing a soft note with low velocity controlling volume, do you want that low volume being saved with your patch?
Or you have a joystick for live tweaking, that is also good to have independent of your edit buffer.
I might start with some experiments using a MB_NG or MB64e. I recently found out TK has implemented a way to send NRPN with meta events...
But it´s a long way till i get to this, currently i am tweaking my analog voltage controlled SID channel and that is still a lot of stuff to fix and callibrate.
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in fact the 2164 is really one of the easiest destroiable chips i know. Next in line from my experiance is LM13700 and SSM 2018...
You can wreck it by simply touching it if you are carrying static charge. And most likley you are... It is good advice to ground yourself (eg by touching a grounded metal case of some equipment) before handling the chip. Don´t forget that this also counts for components that are connected on the pcb. I killed more than one of these chips simply by sticking a resistor into the breadboard while experimenting.
Since i do the grounding trick before touching a part like this i never had any issues.
There are a couple of sellers on ebay who still offer the ssm2164 or the coolaudio replacement.
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So do i understand that you want to remote controll your sid because you are not happy with the encoder speed?
It´s the same for me. I have to turn encoders like 20 times to sweep through the whole filter range.
Unfortunately MB_SID does not provide any different speedmodes like the MB64e.
A few threads down in this forum TK explained how to change the Shift button so that it´s function on encoder speed is reversed. If you try this, please tell us how it went. For me changing that line of code unfortunately did nothing (strange).
I am considering diving into the MB_NG world to provide controlls for a couple of features. Analog pots sending NRPN would be my choice.
Freq, Reso, 2* ADSR and depth, EXT_CV 7, EXT_CV 8, maybe even direct controll for LFO 1-3 are the controlls i am thinking about.
combined with some encoders from the original cs this would be killer for fast tweaking of sounds!
But i could not try out how this behaves yet. EG, if a NRPN message will automatically be routed to the selected Sid (1-4, L, R, LR) no matter what is set in the ensamble.
I kind of fear that building such a "programmer" would require to read out the ensamble channel settings from the sid, check what sid is active on the cs by looking at the pins driving the sid leds and then decide what midi channel the data has to be sent on. That shure would be possible, but rather complex for a programming noob like me.
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You could also consider to use the first two outs for mixing. Channel 2-4 would then need a dual switch to connect to the mixed output. This way you can easily change between using individual outs or a mixed output without replugging any cables.
Try these old opamps if you like the sound it´s fine. YOu should check with a scope if these start to oscillate when driving cables. As for me, i would rather use those in a distortion effect of some kind rather than using them for main outs of any device.
You want something that is capable of driving any real world load easily. 5532 is a good choice as it´s cheap, low noise, stable under usual conditions and can drive 600r loads. So your sound will be uninfluenced by cable length and the load it sees.
Of course TL072 is an option often seen, but it is much weaker so the outputs are more prone to picking up noise and hum compared to the lower output-impedance chips like the 5532.
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Your schematic looks fine, although you have to consider that the ssm2164 will be at unity gain with 0v and attenuate with a cv going positive. around 3v should be enough to completly shut off the signal. So your cv needs to be inverted (easiest way is by inverting it in the ensamble settings) and scaled down. I don´t know if the trimmer on the Aout module is sufficient in it´s range for this (i am driving my vcas with the dac directly, TLV5630 outputs around 5v and can drive 2k).
If you cannot scale the cv down to the needed range with the trimmer you need to add an aditional voltage divider in your cv line. Like the trimmer in my drawing. Impedance of the divider can be anything from 10k to 100k, the VCAs are truly voltage controlled so impedance doesn´t matter. You could try this :
Aout --- 10k --- SSM2164 cv in
|
10k
|
ground
this will reduce the cv from the aout to one half of it´s original range. Then adjust the trimmer with cv at full level (volume=0 when aout channel is inverted) until the sound is completly vanished.
If your cv range is too high the notes will be delayed as the envelope will attack through too much silence in the beginning.
If the cv range is too small you will still hear sound even though the note should be turned off.
I left the mode pin open. For a synth neither distortion nor noise in the order to be expected will be of any concern. I doubt that you´ll hear any difference.
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same for me. integrating mixing is very convenient. Although i have a 40ch mixing desk for all sorts of tricks, it´s sometimes nice and handy to be able to have all sids mixed into a single out from the cs.
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Yes, it should work. I have a mixed setup as well, but with a selfmade psu.
+-15v get regulated to +-12v and + 9v
5v is generated from an additional 9v transformer. Current draw is too high for making the 7805 drop some 10v if you use an lcd backlight and maybe additional slave cores and lots of LEDs.
Might even be good to have a 7805 for each slave core.
Make sure that you don´t connect 9v to 12v! 7809 can be driven from the 12v line with no problem. Current draw should be rather small, but still it´s good practise to use a heatsink.
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from the cc implemention chart (sid manual):
The CC implementation covers the most important parameters at 7bit resolution, so that they can be easily accessed from a DAW or a common MIDI controller. CC changes are non-destructive and won't affect the edit buffer. This isn't a bug, but a feature! Means: your edits on the Control Surface won't be overwritten by incoming CCs, but as an effect they won't be displayed on the LCD, and you won't be able to store changes made via CC as a new patch.
I still have a lot of old pics here, so if anyone has a hint how to make it send NRPN on pot movement that would be great! In 8bit programming section there is an example for sending cc on pot movement (first example in the list). It is propably necessary to make it look what sid is currently active (simply by looking at the dout pins driving the sid 1-4 LEDs), so that data can be sent to each connected core. Not sure how this has to be implemented. Propably hard if different ensambles with different midi channel mapping will be used...
In this case it´s propably best to have the controll pic handle midi mapping and use just a single ensamble on the master sid.
What i am thinking of is a box that will controll around 10 to 16 parameters from analog pots in full resolution. I was nevery happy with the feel of encoders for tweaking the synth. I am a studio user, so i wouldn´t mind the parameter jumps. And there is still the option to use encoders alongside.
my analog sid channel - vc wavefolder, FM, overdrive - sounds and schematics
in MIDIbox SID
Posted
does anyone know if it´s possible to put audio examples in the forum post as miniplayers instead of downloads?