Renoise midi clock to control voltage?

I’m working on synth project using a Raspberry pi to run Linux and i’d like to use a midi clock from Raspberry to sync with different analogue modules.

I can’t find a cheap and simple solution for this, though i guess the DoepferMCV4 will do the job, but how do i make Raspberry pi feedthe midi clock signal tothe MCV4 with? Do i need specialized scripts/software? Would it be sufficient to hook it up with a USB to midi converter? Are there easier/better ways to get around this?

Is it possible to connect a sound card to raspberry pi and then use software to convert the midi clock to a CV signal through the audio jack? I only need the cv for triggering/syncing purposes, not for pitch controlling.

Any other ideas?

Note shure if i follow, but i guess there should be a way to connect a simple cv gate signal through some of those pins, though i would assume i need to know a little coding if i want to make this work. I really hoped i could avoid the coding bit as i don’t have any real experience with it and i’m not very motivated to start learning it.

I found a linux app called “jm2cv : Jack Midi to Control Voltage” which i assume could do the trick and i assume i could use jm2cv to feed a CV signal through a sound card? Does this seem right or have i misunderstood how this work? Could i tap this cv signal from a midi out from the sound card or what kind of connector could i send an analogue signal like this? If it only goes through the jack out/headphones then would i need two sound cards and would i be able to use two sound cards at once in Linux at all?

I guess usingMCV4 with a raspberry pi soundcard with midi output is the simplest solution, but i’ll try the alternative first to see if i can make it a lot cheaper for myself. It would be a bit ironic if the midi to cv circuit would be the most expensive part in the whole machine. It costs even more than the 7" touch screen and raspberry pi 3 with ssd module combined and stillMCV4 is considered cheap. Total ripoff imo, though i guess these don’t sell in enourmous quantities and are built for professional use, so i guess that’s why they cost as much compared to what you get.

Another question: would a raspberry pi 3 be capable of running the latest ubuntu distros or even Renoise at all? I would guess that is to stretch it’s capabilities with its 1GB RAM, but it seems it has a pretty powerful CPU.

It’s not crucial that it can run Renoise though, but it would be cool. I was hoping i could at least run milkytracker or some other less cpu and memory hungry apps to generate some input to the crazy machine i’m building.

Plan is to use raspberry pi as a sound source and clock for two 8 step sequencers and several ISD (audio recorder chip) modules and make them act like a sampler, maybe even some echo/short delay modules and ‘voice changer’ module based on a HT8950, the same one used in ‘Synthrotek Roboto’ synth module. All analogue except from the raspberry and with fancy lighting switches. :stuck_out_tongue:

Edit: Ah, now i see there is a ubuntu distro called Ubuntu MATE, especially made for Raspberry pi. I can already smell the headache it’s going to be setting this up… :confused:

Sorry Bellows, I understand better what you’re getting at now, but realistically, if you want to be able to control stuff using hand-built gear, it’s likely that coding will come into it at some point. That said, I wasn’t aware of that software you just mentioned, and that sounds like it might do the job, but I’m not so sure you would feed a voltage through a soundcard for this stuff. I think it probably uses the expansion pins to send a voltage cause that’s kinda the point of a Pi, it being an experimenters board/computer they give you the expansion pins to play around with.

It’s an interesting idea, using the soundcard to transfer a voltage to another, but I’ve never heared of CV being done that way before. CV might use the same jacks, but what’s behind those jacks is different to a soundcard, and I can imagine the levels would be drastically different as well. I suppose there’s nothing to stop you sending a ‘tone’ from a soundcard and creating a little circuit to receive it and convert it into a clock voltage though.

I’m using several analogue modules that uses chips with a clock pin, so i’m pretty convinced i’ll be able to control their timing in some fashion using a CV gate signal. The audio jack should be very well capable of delivering a 5V gate signal. It doesn’t even have to be 5V as it reacts to voltages even below 2V. Actually if i was able to route a specific track directly in mono through a sound card line out(or rather headphones out or else i might have to amplify the gate signal to make it usable), i think i could generate the signal using a square wave LFO hooked up with a DC Offset device sent through the headphones jack (have i misunderstood how this works perhaps? Is there a DC filter except for the tick box in the master post mixer?). It shouldn’t be too hard at all if the softwares allowed me to do this. By connecting the gate signal to a decade counter (4017) i could easily get 8 new gate signals that turns on in sequence in timing with the original gate signal. I could perhaps send this gate signal using Renoise, but then i assume i can’t route audio from renoise to the machine at the same time unless i go mono, by using one channel for the gate signal and the other for audio. I can also use the gate to trigger on the negative phase, then i could hook it up with another 4017 and get 16 steps, but for this project 8 steps will have to do. It’s not supposed to act like a normal seuencer anyway, why would i need that when i have Renoise? No this thing is for making weird sounds and rythms, hopefully a bit more than just the average noise box, but more or less that is its purpose. The project is a bit based on the concept of ‘less thinking, more building’. :smiley:

If syncing computer with the analogue is too much of a hazzle i’ll just make a simple clock with a 555 timer, they’re always fun. :stuck_out_tongue:

If you want to use a MIDI clock for controlling various analogue equipment, then you have to convert the MIDI to CV, so i don’t see what’s different in this case? The normal route from a computer would be from a MIDI output of the computer sound card, wouldn’t it? I know doing it through the jack is maybe not the usual route, but i basically just wanted to know how jm2cv works, does it send non MIDI signals through the MIDI connector or does it send it through one of the jack outputs? I can’t see there are any other options and to me it sounds like sending it through the MIDI connector would be a bit strange because where do you find a MIDI connector cable to a single jack? I can’t recall seeng one of those around lately.

I’m not sticking the 5V into the input of the soundcard, this is what goes out from the soundcard and a regular headphone output indeed delivers about 5 volts or else it wouldn’t be able to drive the passive hedphones. There is quite a bit difference between the unamplified signal that goes into the preamp than the signal that comes out from the power amp, which a headphones output is. :wink:

The line out however, is a weaker signal that has just gone through a preamp circuit so it becomes a solid enough signal so that it can travel through longer cables and still deliver a decent output to whatever line in you put it into.

I basically just want to send a 2-5V low frequency squarewave signal through the headphones jack, thats it. However i’m not shure this works, because i have a hunch that soundcard might have a DC filter, because audio equipment aren’t particularly happy with those. Doesn’t it always say 20Hz-20kHz on the outputs, i can’t recall seeing a 0Hz-20kHz or whatever?

Also, milli stands for a thousand to one, not a million. I believe most line ins could handle an audio signal that peaks around 5V for a short time, you would most likely turn the volume knob down to a much lower level pretty soon when you hear how it sounds like anyway, but feeding it a 5V DC signal is most likely not a good idea.

The thing is i know a bit of analogue electronics and i have a fair understanding of how each component works and can read schematics well enough to solder together basic stuff, but digital electronics however is a completely different world that i have next to no experience with.

Edit: that MIDI bit thingy looks like the exact thing i need and at a more decent price tag, but it strikes me as a bit weird that it gets all the MIDI signals through a 3 lead stereo jack. I thought MIDI used more than 3 leads, what’s the point with the weird din plug connector then? Or perhaps the 3 leads covers the most useful signals?

Well, like I said, you’re way ahead of me with this, I must have been getting confused with watts :unsure:

It’s a shame you’re not documenting the creation of this synth you’re making, or maybe you are?

Do you have a blog or something like that, something you update as you build it?

Hands-on sampling is surely pretty high on just about every electronic musicians list, or at least it definitely is on mine, you only have to look at the popularity of the Volca Sample (and that thing can’t even sample).

Would love to see this thing progress, or if you can’t be bothered with that, maybe do a tutorial when you finish it :slight_smile:

Voltage multiplied by current = watt,

most electronic circuits are driven by direct current, DC, while audio is always an alternate current, AC, because that’s what makes it audible. As long as the current is alternating in any frequency between 20Hz to 20kHz it shold be audible at least in theory. 1 Hertz is exactly 1 cycle per second, 2Hz is 2 cycles and so on. The voltage tells us how ‘loud’ this signal is, but is impossible to measure with a voltmeter because it’s constantly alternating between negative volts and positive volts more than 20 times per second while the voltmeter tries to read it as DC voltage.

In circuits there is always a common ground, which is supposed to be 0V at any time, you can look at it as a huge water tank, though in circuits we usuallly read it from positive to ground, or negative to ground in other circuits (both in opamp circuits used in many amplifiers), but in fact it actually is the other way around because electrons are negatively charged and therefore is drawn towards the positive leads. A negative voltage however is basically just to swap the wires from the power source.

At this moment i’m in the experimentation phase and i’m basically just waiting for all the parts i need to start this. I’m building most of it out of small ready made modules this time, instead of building them from small components, so this simplifies things a lot compared to soldering everything by hand like i have done for the most. I’ve ordered most from Aliexpress and they’re mostly pretty cheap compared to what they’re capable of. The most expensive parts was of course the Raspberry pi + extras, but apart from that i have ordered things like a tube preamp for about $10 and i’m most certainly shure it would cost me a lot more to build myself. Surround sound modules (i think i can easily turn it into an echo/delay module too), tone board, preamps, voltage regulator modules with voltage output display, recording modules, ring modulator kit, robot voice kit, relay module, radio modules, radio transmitter modules, i’ve lost count of all i have ordered, but the price range is somewhere around $3-$5 per module. I’ve also ordered metal switches with built in light when it’s on and lots of potentiometers, wire connectors and such. There will be lots of wiring, but i think the project will be pretty easy to explain how it all works, so i will try and document it as well as i can and post a thread about it once i’ve figured how and what i can get crammed into this thing.

My current plan/configuraton is something like this:

Raspeberry pi as an audio generator, hopefully Renoise feeding audio through a soundcard > signal goes to a preamp and mixer which splits the audio signal > each signal is fed through a recording module and dry signal to final mixer > 8 step sequencers triggers the recording modules to play > various effects like reverbs and more crazy stuff like pitch/robot /vibrato voice changer which is easy to circuit bend into sounding completely bonkers

Only a crude simplified explanation, but it gives you an idea of what i’m trying to make. All recorder modules can record what comes from the soundcard by holding a button and you can also change sample rate by turning a knob (or one for coarse tuning and one for fine tuning) after i’ve modified it a bit. The sequencer, i hope will work with an optocoupler connected to the play button which can be set to the mode where it plays the whole recording or just while it’s switched on. It also has a loop mode so you could basically record a beat, and make it loop forever and also turn the knob where you can make the bpm match the rest if you wish.

I doubt i will use the radio modules for this project, but anything can happen. My initial plan with the radio and radio transmitter modules was to make a theremin out of them, but i’m not shure it will work.

The cream on top would of course be to have Renoise on the touch screen and at the same time have it synced to everything else. :smiley:

Good to hear you plan to start a thread, really looking forward to that, Bellows!

I look at a lot of these DIY projects and they’re all pretty neat in their own ways, but you know what always seems to be the part people turn chicken about? It’s when it comes to incorporating analogue stuff. Like for example, you’ll see DIY samplers but they leave it at that, just digital playback of a file with a trigger attached. They don’t bother adding an analogue filter and envelope, so that the envelope gets triggered along with the sample playback of the sampler, and that’s a real shame, cause lo-fi samplers sound amazing through analogue filters!

I don’t know if there’s any part of your plans that incorporate it, but if there’s any part that uses digital control of analogue filter and envelope, that’s the part I would personally be glued to with interest cause like with most people, that seems to be the barrier into getting a little more advanced. Again, it’s a real shame it’s a barrier, cause the Roland synths of the 80s, for example the JX-8P, were digitally controlled analogues. It’s a powerful combination and even allows for presets on analogue due to being able to store the digital values that are controlling the filter and envelope etc.

I’m afraid i will not make digital control for the analogue parts in this project, but it would pobably not be too hard to make digital controls if you use Arduino, Raspberry pi or similar. I would assume that you could simply buy some digital potentiometer modules and use them in place of the mechanical (regular analogue) pot meters that i’ll be using. I haven’t worked with these yet, but i assume they work identical to regular pots, just that instead of turning a knob you instruct its value through the arduino/R pi. In software almost anything should be possible, but that’s definately not my field of expertise, so i have really no idea how you can make it work with different softwares like Renoise or whatever.

If you’re interested in digipots you should take a look at these articles for R pi and arduino:

My biggest worry with these is that i believe they could introduse noise if the input signal isn’t 100% stabile/accurate, but i don’t know if this is gonna be a problem using r pi or arduino.

If you are going to use analogue filters for digital audio you have to remember you also need a DAC(output of a soundcard) and then you’re left with an analogue signal and if you want to turn it back to digital you need an ADC(input of a soundcard) after the filter. Both these steps shape the audio to a certain degree, depending on the DAC/ADC circuts/algorithms. You would end up with a digital filter with analogue filtering.

Yeah, you should definately start figuring out those things and maybe teach me some cool digital stuff.

Here’s my last attemt at a sequencer, it works well, but only outputs square wave output with no envelopes. It has some echo/reverb stuff too. Doesn’t really sound very pleasant, but very fun to play and annoy neighbours with:


It is an 8 step sequencer, but capapable of sequencing in several different step settings and i’m hoping i can make the one i’m building now to work similarly only with samples and hopefully some other weird stuff

I ordered some better recording modules today too and i can’t complaint on the price, i paid $28 for 8 modules. These are easier to add pitch shifting to also, it has it’s own pins for it, also better quality sound and larger memory.

I’m receiveing so many packages these days that i bought for this project i lost count, picked up 8 today and everything is dirt cheap. I wasn’t aware that many came as unsoldered kits, so i guess i have to get on with the soldering. At least now i don’t have to find all the parts myself and use universal veroboards to solder them too, it’s always a headache when you do a mistake. It’s not that hard when you have the specially designed pcb for it, just pop parts in place and a swing with the soldering iron.

wow, now i’m rambling…

Ramble away, your rambling had me on eBay last night and I bought a couple of those ISD1820 Recording/Playback modules, I bought two!

That sequencer looks very industrial, built like a tank, but the burning question I have is does it transpose when you change key on an attached MIDI or CV keyboard?

BTW, what do you mean by “better recording modules”, I’m assuming you used those ISD1820 types before and have found a better one now?

I suggest you go buying your modules from Aliexpress instead, as they’re cheaper over there and i have never yet received a faulty item after more than 150 purchases over the years. It can be a bit difficult to find the best deals and the exact product youre looking for, but if you want i can give you the link to some of the modules i bought.

The better modules: Still cheaper than one of the 1820 on ebay.

It doesn’t come with a speaker, but that is complete crap anyway, so i wouldn’t even bother. This one is more suited for an Arduino project too, i think you can even transfer waves to it from a computer, though i haven’t looked into that yet.

The ISD1820 module is a bit more easy to work with in an analogue circuit though, because they have different pinouts, analogue pins on the ISD1820 and digital pins on ISD1700.

Both chips should be capable of recording in 12kHz, but i think the default rate is lower on the 1820 module, though i’m not really shure. The 1700 has a longer recording time and is better suited for digital control, but can also be used for analogue circuits by using the pinouts from the microswitches.

Edit: when looking closer at the 1700 module i can’t find the resistor pins for changing the sample rate and i realized this module is a bit different from ones i bought earlier.

If you got this version of the 1820 module then pitch shifting is very easy, just replace the jumper above the playl button with a potentiometer:

If you don’t have this version you will most likely not have this pinout, but then you will see a resistor that goes from pin 10, in my case it’s R4 on two different modules, so i guess that should be it. This resistor is connected between pin 10 and ground and it should be a 2,2Kohm resistor. You need to find a way to connect this 2,2K resistor to a potentiometer, 50K should work fine, to ground.

Options would be to remove the existing R4 and solder a 2,2K and 50Kpot in it’s place direclty from the 10th pin under the board and connect it to any ground lead. You could also scratch off the copper that makes a short to ground on the one end of R4, solder a 50Kpot from this lead to any ground lead/pin.

Damn … I was going to go for that one due to it having the switches where the other board (the one I purchased), uses jumpers instead. I went for the one with jumpers thinking it would be better to connect to other things (directly to the jumper pins), and it also came with, granted, a crappy speaker but also a battery holder. Didn’t really need either of those but went for it due to convenience really, and I wanted to battery-power them while experimenting with them anyway. The one I went for also mentioned something about being able to increase the sample rate by using a different resistor. Actually though, I’m assuming the one you show does exactly the same cause it’s the same chip after all.

Thanks for pointing that other one out, but reading the spec aren’t you concerned about that indicator feature the board has?

Wouldn’t that get in the way of doing looping and stuff by doing a beep on each repeat of the loop?

I thought i replied to this yesterday, but i must have forgotten to push ‘send’. It sounds to me like you have gotten the perfect ISD1820 module for diy projects, pins are good, means less soldering.

Can you give me the link so i can check it for you? I can tell you what you need to do to make a pitch controller on your module. On the ISD1820 it’s always pin 10 resistor that controls the sample frequency.

I can’t recall any clicks on the loop mode. My modules are mainly (possibly exclusively) gonna use the playe (no sustain after trigger signal) and playl (plays whole sample on each trigger(sustain)) because i use a sequencer to trigger it.

I’m hoping i can simply turn a 8 step LED sequencer module into a 8 step photocoupler sequencer directly connected to a switch that selects between PLAYE and PLAYL. I’m not entirely shure if it’s possible though, because i’m not shure the photoresistors can go to a low enough resistance. If it doesn’t work i can either try to make a transistor switch circuit or use a relay instead of the photocouplers, or maybe use phototransistors instead of photoresistors.

Reading the sheet he included, I’m a bit weary of the way looping would need to be handled. It looks as if there’s no on-board way of setting it to loop mode, which is kinda dumb really. According to the sheet, I’d have to reset a pin each time I want a loop. That makes me think I’d get a click due to the reset on each loop. Looking forward to playing with them though!

Reset a pin? Couldn’t you just use a switch for this?

That looks exactly the same as mine, except from the fact that the components are labled differently, but they have used the excact same layout, so that doesn’t matter.

It looks like it is just bad labeling, those to the left is for the play modes, while the P-E to the right is the loop pins, if you put the jumper there or a switch you can have an endless loop of whatever you recorded.

Thanks Bellows!

I downloaded the official Datasheet for the chip and was reading it yesterday. Have to say, I don’t like the idea of it having a built-in amplifier with no way to bypass it. Wouldn’t the levels need to be taken down before connecting it to something like, say, a multitrack recorder input?

Having an amplified output would mean it’s way above line-level, wouldn’t it?

Still learning this stuff but I can imagine the distortion would be insane!

It’s a lot better to have an amplified output than a too low output. The signal out will be about 5 volts at max output, while a line output is around 1 volts and if you feed a signal that peaks at 5 volts into a line in, then you would most likely end up with a lot of distortion/clipping. However this is very easy to fix with some resistors/potentiometers.

In some cases an amplified output that is too loud is very beneficial, like for instance if you want it to go through a passive EQ circuit. A passive EQ circuit is pretty easy to create, but it requires a louder signal than an active EQ circuit (which basically is a preamp or several preamps and a passive EQ circuit). An EQ circuit is basically just some potentiometers and some capacitors.

If you are going to use the signal to feed several line inputs it will also be beneficial, in basic you only need to add a potentiometer to each input.

Think of the potentiometer as a mixing device, it’s got line in, which is the middle taper that moves across the graphite (the middle pin of the potentiometer). It’s got a line out, which goes to the line in of whatever you are feeding the signal into. It can be on either side, it’s up to you, but the usual way would be to turn up the volume clockwise. The last pin goes to ground. If you look at the potentiometer from the top, pins facing upwards, it would be pin left to ground, middle pin is line in and the right one is the line out. You could also swap the line in and line out, it doesn’t really matter. Find the way you like and stick with it, it’s easy to solder the wrong way and you realize just when everything is screwed together.

You might need an extra resistor to prevent it from going too loud, but that depends on what you have in the other end. It could be wise to use a trimpot, maybe 50K and preferably multiturn pots so you can finetune it easily at any time from 0 to 50K which should be enough. this one shold not be connected to ground, as that is the first potentiometers job. If you use a small value on the potentiometer it will act as a filter and remove higher frequencies, so it might be wise to use higher value potentiometers such as 500K. Preferably it should be a A500K which has a logarithmic curve. If signal has zero resistance to ground, then it’s completely lost and should in theory be completely silent.

Thanks, I even understood that!

I’ll have to get the electronics stuff out again, cause I shoved it aside while I was doing other stuff. Last thing I was trying to get my head around was impedence, so I’ll carry on from there I think. And I was looking into the most basic way to convert DC into AC cause I read that a lot of the DIY synths out there are outputting a DC voltage, but ideally, the voltage should be converted to AC to drive a speaker.

Impedance is quite complicated and isn’t something you need to worry about too much as long as you use circuits that work properly and is designed by someone who knows their shit. Just about any proper audio circuit has a high input impedance and a low output impedance and as long as you use the proper power supplies and proper grounding, and everything then it should be fine.

I think the problem you’re referring to is that audio signals from crappy circuits can have DC offset and clipping problems.

You should never play DC through your speakers. DC is inaudible except from the initial click and the sound of your speaker frying. Audio is AC and should not have any DC at all, but if a DC is introduced to your audio signal creating a DC offset then you probably have some bad circuitry causing it.

No I meant that I was intending to play around with basic oscillators and driving a speaker etc. I started there mainly because I wanted to be able to hear the results of experimentation. I can’t remember which one it was, but there was one where someone showed a basic oscillator circuit but pointed out it’s not ideal for driving a speaker due to it being DC, and crazy as it might sound I’m pretty sure he said it lacks a current. He mentioned something about using an opamp to create a buffer that would either convert the output to AC or give it a current. I’m pretty sure it was one of those two reasons but can’t remember which. It might even have been for both reasons.

So that’s what I was looking into at the time; trying to get my head around what an opamp buffer does and to understand impedance.

When you oscillate a DC signal it’s not DC anymore, so how can an oscillator circuit ‘be’ DC? Almost all small circuits are driven by DC, but an audio signal is always AC, it can’t be both at the same time. If the audio signal has DC offset however, the voltages rises while volume/dynamics of the audio decreases, which is not good. Bad circuitry can of course cause things like this, so to avoid it you should stick with circuits you know works. Ready made modules makes this easy, while building your own from random circuits you find on the net however can often lead to problems.

An op amp is a common integrated amplifier that increases the amplitude, it’s basically a preamp on its own. It however is driven by both positve and negative voltage sources, which makes the circuits a bit more intricate and makes it harder to figure out how to make them interact properly with different circuits. Ready made modules should have all this sorted out with only one power supply, so that makes things easier.

I wouldn’t use opamps for oscillating, i’d rather start with 555 oscillators, much more fun to play around with. Then i’d buy something like a function generator IC, i strongly recommend XR-2206, it’s one chip that can produce all sine, triangle and square wave and it’s super easy to work with, just build one of the circuits in the datasheet.

One cool suggestion would be to make an LFO out of a 555 timer chip that controls an LED, then turn it into a photocoupler and let the output of the photoresistor control the volume of the XR-2206. To make it even cooler you could connect an IR-phototransistor to the control voltage pin of the XR-2206 and hook up an IR LED to a power source, now you can control the pitch with your hand like a theremin only with IR light.

It might be a bit inaccurate to put it that way, because even though it’s an AC signal it can still be very close to a DC signal. A square wave oscillator is in fact DC only in pulses. If played in high enough frequencies it becaomes audible, but below the audible frequencies it acts like a continuous on and off DC switch. DC is when the audio is like a flat road while audio is like the ocean. Audio should always be centered at zero or else it has DC offset which can cause clicks, distortion, weaker volume and all that crap. The further away from being sentered to zero the worse it will sound. You can fix it with a highpass filter, but it may already have destroyed the audio.

If you want to make an oscillator for actual music/audio production, then i suggest sticking with a function generator module or the chip i mentioned, as you won’t have to deal with the problems from bad ciruitry.

I think i will order a cheap oscilloscope thingy soon, it could be very useful to see how the waveform looks like. I saw some ridiculously cheap ones at aliexpress, but i’m not shure…

Can you just clarify my thinking here for me, Bellows?

Looking at a signal on an oscilloscope, I have it in my head that a waveform that falls either completely above or completely below the center line is DC, and anything that has the line running through the center is AC. Assuming that is correct then what you’re saying is that when something deviates further and fruther from the centre is gaining a larger and larger DC offset, right?

BTW, that chip looks cool, I started looking into it but I found another which appears to be even better cause it has Saw and pulse width modulation whereas the othe doesn’t. It’s called ICL8038 if you fancy taking a look, I think it’s the same sort of thing, looks like it from the spec sheet. Regards the oscilloscope, I actually have the same model he uses in the videos although I’ve not built mine yet, and I purchased it as a kit! Have to say though, I’m pretty sure it’s a decent piece of kit as lots of people use them and I’ve never seen a bad review about one. It’s not as good as having an analogue oscilloscope would be, obviously, but for the money it appears to be quite neat!

Sound just about right. The simplest way to put it i think is to say that if the sum of the negative and the positive phase is zero then you have a DC balanced signal.

I haven’t seen the chip you mentioned, it’s datasheet says it’s obsolete, but it looks like it’s still being sold. I ordered a couple myself to check them out.

Another chip that looks interesting is the SN76477 (orICS76477) , an old chip used as a sound generator in arcade games and such. It seems to work like a complete synthesizer in one chip, with envelope generator, LFO, noise generator and oscillator.

Didn’t realise it was obsolete, lol, never saw that (I ordered four of the things) :rolleyes:

Will check-out the others you mention although it would be much cooler to know about current production chips that are designed to be complete synthesizers if you know of any!

Thanks for the heads-up on the AC/DC thing :slight_smile:

They don’t make much of these kind of things anymore, digital has taken over.SN76477 isn’t that expensive atm, so you better get some before they become rare.

Analogue is still used in guitar pedals and such and it’s always going to be a part of the circuitry in musical equipment, but in the whole picture it’s just a small niche.

It’s not that expensive to reprint an IC if you already have the masks for it, but don’t expect new analogue ICs to be developed and germanium transistors are almost obsolete and very expensive.

Lots of older parts are much less accurate than the newer parts that has taken their place. Accuracy is not the reason why we want to use analogue equpiment for audio and if you’re redesigning an old circuit with modern replacement parts it won’t sound the same. For instance i believe you can make a better simulation of a germanium based fuzz pedal digitally than if you’re building it with silicon transistors.

I’m no expert on the subject, so i might be wrong and i’m not on the level of making really high end musical equipment as it’s just a hobby and i want to make it cheap and simple.

I’m still hoping for a new stock of the6581/8580 SID chip (C64), there is a replacement chip made for it that digitally emulates the SID, but it’s not really what i’m looking for.

The chipSN76477 is the closest i can come to the SID that is cheap and easy to find stll.

When i talked about new vs old components it’s mainly the transistors, diodes and integrated circuits that has changed. Some sellers, selling old parts, advertise with ‘better sounding resistors’ or whatever, but that’s just bs. Maybe you can hear some difference between different types of capacitors in some circuits, but you could hardly say that one sounds better than the other.