Reassurance Requested for a New SSP Build

“Sale ends in 27 minutes”

I think I’m going to miss out on this deal hahaha

Thanks for the heads up though! I’m still exploring cases, external modules for integrating other line-level gear, how to buy and if I should purchase these things now or later as I’m in the EU until the end of August when I go back to US.

Looking at the new MI Veils because it has 20db of gain so I can use it as a quad line-level to modular-level converter, or I can do the DC offset CV conversion for pulsar/ssp conversion. I could also patch the 10V pin from the pulsar to the first input and use the normaled outputs of each output as control faders to give the SSP some more knobs. Seemed pretty efficient for 10hp.


I just wanted to make some clarification on Windows and using the SSP as a audio interface / audio over USB interface.

You’ll have to use a DAW with WASAPI support, or otherwise you’re left with Asio4All driver in order to connect your SSP to your Windows. But even then, you could run into problems.

I’ve tried Ableton, Reaper, FL Studio and Bitwig, and I’ve tried Asio4All, Asiolink, WASAPI, MME… and a couple more drivers, but I have never had a pop free and click free audio streaming between my Windows and the SSP.

The only solution to my problem was to use Mac. The Core Audio driver does an excellent job.

My windows experience was very bad, I can absolutely not recommend it. I’ve tested the SSP with all DAW’s and all available drivers 3 different computers, all Windows 10, none of it worked as I expected.
I couldn’t stream a sound for more than even 20 seconds without clearly audible pops in the audio.
Even the CV I’ve sent from DAW’s to the SSP started to bug out (LFO sinewaves become glitchy etc).

My advice - get a cheap Mac M1 when you want to integrate DAW and SSP flawlessly. Or maybe also Linux, but I have no idea about that.

But in case you get your setup to run smoothly with Windows, let us know how you do it!

Oh no that’s heartbreaking, all I’ve got is my windows 10 laptop. Were you ever able to simply record audio over USB into any sort of DAW on Windows (without clicks or pops at a standard 24bit 48khz rate?) I’ll try it out and let you know how it goes.

I guess I don’t need to send audio from Windows to SSP, but I’d really want to be able to record if possible. If I can’t send audio I’ll just hook up my SSL2+ interface and pipe the audio and midi in and out of that hopefully.

Edit: I just remembered I could do internal recording and then move all the wav files over to my laptop at a different time, so this isn’t as much of a problem. If they get recorded to the SD card then I could SSH into the SSP and SCP the files over instead of recording them directly to the DAW. The pull factor of this thing is the acronym usage.

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Roland System 500 Eurorack Case SYR-E84
It’s 84HP, it’s got a lid, it’s metal, it’s available, it’s affordable for me, I can screw two of them together to expand it in the future, 2000mA for the +12V rail so if I get these modules + 500mA to power Deluge = 1612mA at a maximum, that gives me great headroom, portable as well

Mutable Instrument Veils
10HP, can output 8V CV to act as faders for SSP for easier control, can add 0-8V offset to convert the -5/+5 volts from SSP to Pulsar 0-10V, I need more VCAs because I use the only two on the Pulsar, can run line level into Veils and boost +20db (or 30db as it says in the manual but forum people say 20, enough either way), I can run eurorack audio into Veils and attenuate down to line level, I can use it as a mixer or stereo/mono line/euro audio level converter. One thing I don’t think this does is a negative offset, so I don’t think I could bring the 0-10V pulsar CV (and audio operates at 0-10V on pulsar according the to manual, but maybe the mix out takes out the DC component? Might bring it into work and look at it with an oscilloscope or VU meter…) down to -5/+5 input for the SSP, but I don’t think it matters to me. I think the biggest impact would be SNR for audio into SSP would be “less than optimal” because I’d take a 0-10V analog signal, go down to 0-5V to prevent clipping, then digitally amplify the signal with the introduced sampling noise as well up to the -5 / +5V “amplitude width”, but (if I even notice it) I could use the Listen I/O to bring audio levels up appropriately for higher quality sounds if I need to. Probably overthinking this.

Listen I/O
I’ve used it before and it works perfectly for what it does. Line to Euro and Euro to Line, The line outs both act as headphone TRS outputs which is super handy, acts as a mult or mixer as well if you really need it to. I’ll use this to bring Deluge up to Euro level and then SSP audio out to headphones which frees up the Veils for fader, VCA, fx loop, mixer duties

the main event

8HP Blank Panel
Most important, I’ll put a sticky note on it that says “Practice”

I’ve got to check if I’ve got the correct USB cables and pick up that ethernet SSH enabling LAN cable thing in the other thread. All things are generally available in stores, gunna wait two weeks to take care of some things then figure out the best way to purchase things. Looks like House of Sound in Switzerland has all of these things currently in stock which would be easiest, but there might be cheaper options if I look around a bit more. Let me know if you’ve got any thoughts or recommendations! Would synth shops haggle with me if I asked them? :money_with_wings: :slightly_smiling_face:


recording from outside the box works just fine!
Its only if you send audio and CV modulations to the SSP from your Windows that can cause these pops.

I figured that 24 in and out channels are too much for Asio4all.

But in my experience, routing a oscillator into an input on your SSP to then record it on Windows should work perfectly fine.

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I’ve committed, won’t be able to get to it for a month or two because of life circumstances, so for now I’ll start learning some JUCE framework.



Your in for a lot of fun when the time comes, the SSP is a phenomenal piece of work and it sounds fantastic too, so versatile, the contributions from others and in particular @thetechnobear are outstanding, a personal thank you to him (first post here) for all those modules you built and give to us, spoilt (-:


Hello all! I finally got back home and have been learning my way around SSP. I’ve sunk my whole morning into playing around with the granular engine and I’m having a blast. Had two questions someone might be able to answer.

Can I add notes to the manual forum posts? Just like, new user student notes. For example the Granular Module Granular Module - SSP manual - Percussa Forum doesn’t have a section on what the Dub parameter does. It’s probably around here somewhere on the forum, but can I add a comment to this forum post? Would it be desired if I did little updates like that to the manual posts as I found them?

I got a Roland SYR-E84 case, the SSP, and a 4ms Listen module for outputting audio. I’m getting noise on the output. I have the SSP connected directly to channel 1 of the busboard and the 4ms Listen connected to channel 2 via a daisy chain ribbon cable. I turn the output volume on the 4ms all the way down, and I’ve got noise coming out of the output, but if I turn up the output volume a little under halfway, then the noise drops to zero, then if I turn it up to full output the noise comes back (louder obviously). No input to 4ms, no noise, turn down the level and connect SSP output, noise, turn up level to a little under halfway, noise goes to zero. I’ve got the case plugged into a power supply which is then plugged into the wall. I’ll tinker a bit later just plugging directly into the wall and just outputting audio directly to headphones from the SSP instead. My two initial thoughts are that I’m either pulling too much current from the power supply or that the 4ms listen module has something wrong. I don’t think it’s the first one since this power supply can handle more than twice the current that I’m currently pulling. Could be something with SSP, no clue yet. I’ll snoop around on the forum to see if other have had noise issues in the past.

Other than that I’m going to try and build a DFAM patch from scratch and I’ll figure out how to share it whenever I get done with it as my first project to learn my way around the SSP. Thanks for all the help you’ve all given me so far! :slight_smile:

adding notes to ‘SSP Manual’ , sounds like a great idea… why not :slight_smile:

perhaps we should setup up a community wiki for the SSP, it could not only cover modules, but also other FAQ, or patch techniques?

noise via listen…
I dont have the 4ms Listen, but my experience is output modules don’t play nicely wiith power hungry digital modules in the same rack (esp. on same busboard)

I’ve a Befaco Output module, and I had noise from the SSP , but I also had noise from a Qubit Nebulae and my Terminal Tedium… ( my bela salt seems ok)
I think its not so much the power supply handling it (i.e mA) but its just the way the ‘cpu’ switch fast and then eurorack has got these tiny ribbon cables for grounding etc, so it ‘spills’ over.

so generally ( I make exceptions!) , I don’t going into a eurorack mixer/output module, with my SSP, rather I go to either an audio interface, or something like my Octatrack.

note: just to be clear, I find its fine to send audio back to my eurorack for processing… its just I tend not to want to go into a mixer or output module.

all that said, reading posts here, I think mileage varies alot between different users and setups.

as a side note:
Ive generally found eurorack is noisey… its been designed to be ‘easy’ rather than to be clean audio… go to any modular forum, discussing any digital module , and someone will be complaining about noise - and others will be ‘nah its fine for me’
I know, frustrating, but it is what it is.

the only other thing ive learnt with digital modules is to try not to push the gain up too much, helps keep the noise floor lower.

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I’ve got an audio interface, some speakers, and some headphones to try just going direct from the SSP later. I’ll try moving the output module to a different case too just to see what happens. I agree, no matter the product someone will post about noise. In my experience it was always my cheap power supplies whining when I pulled too much amperage from them, or cheap fuzz pedals without proper grounding or noise reduction in the circuitry or something.

Thanks for the ideas!

Edit: no clue how to set up a wiki, but I think the forum suffices for now. I always like sharing notes when/where I can

I wanted to give an update after a month of learning the SSP to finish off this thread. No need to respond to any of my questions, just thinking out loud, I’ll make a new thread for anything pertinent.

Noise issue is 100% from the output module and not SSP, so no worries there. It went away after I moved the module. I get a nifty “Deluge recognized” message when I load up the midi module and turn on the Deluge. I confirmed I can send a note value from Deluge and modulate something on the SSP, haven’t tried doing anything else yet though. The patching took a bit to get used to but it’s going well so far. I put down 16 module then I needed to patch something from page 3 to page 1 and couldn’t figure out how, but then I remembered the Bus module which solved that problem. The wavetable oscillator sounds great, the granular engine sounds great, the sampler, though I’ve only used it briefly works great.

I wanted to see if I could send and receive gates and CV to and from the teletype module. I couldn’t find a way to really generate a sequence of gate signals. I know I can use an LFO but I wanted to skip some steps, and then realized I could use a step sequencer as a gate sequencer by sending out 0 or +5V, and that seemed to do the trick. Then I could receive gates and CV from the teletype via the IN module and I was good to go. I think for my next studies, I really need to get a notebook and write down all my questions. Like, “look up what Dub means”, which I found out later is overdubbing amount.

I tried to make a DFAM as a starting project to learn the ropes, but I couldn’t figure out how to work the VCA really. I was getting tied up trying to get the VCA to open and close with an envelope, and I couldn’t get the envelope to work as fast as I thought it could. I wanted to make a simple AR but I couldn’t get it to be as snappy as I was hoping, but I think after patching some more it was user error. I’m also not sure if there’s just a simple oscillator voice module, or if I should use a series of LFOs for that? I’ll be rereading through all the manuals again and trying again some other time.

I made a new patch and decided rings into clouds is easy and available so I set up a sequencer, quantizer, rings, LFO, and an output module fairly simply and it sounded great. I was a bit tripped up with the quantizer, I went to load a scale and there were like, 3 folders worth of scales that divided the voltage range into exceedingly strange divisions and was wholly unfamiliar to me. I think I found something that looked like a major scale in one of technobear’s folders and it sounded like it worked so I just set it and moved on.

The reverb is absolutely beautiful, I’m in love.

I loaded up a wavetable oscillator and got the classic subtractive synth voice VCO, LPF, ENV, VCA working. I think I’ve got some weird recordings on my SSP, it was a demo model in a store I believe? I’ve got to find some actual wavetable files to load up on this thing, I was just using whatever .WAV I could find. I got some weird plinks with this and tried to figure out how to create a mixer when I realized PMIX exists, so I loaded that up and it’s everything and more I could have ever asked for in a mixer on the SSP so thanks a ton Mr. Technobear for all your hard work. I made an aux send return from there to the delay / reverb combo.

Is there a way to check the CPU percentage? I’ll go snooping around to see if that’s been asked. Since it hadn’t broken yet, I kept building, I used the teletype to vary the sequence clock on the wavetable plinker. I used another teletype clock to gate a sample from SAM. I think it was an elevator door field recording thing. Another sample on SAM I modulated the start and length and set it to loop which gave these ba ba ba ba ba ba babababababbbbbbbbbbbbbbbbbbbbbbbbb noises. I dumped that into clouds then that into PMIX. I got the sound engine to crash after flicking through the clouds modes a bit quickly, like turning left through four modes on clouds and back again. All the audio stopped, I heard an eeeeeeeeeeeee noise, but SYNTHOR application was still working, I could still scroll around and do things. After maybe 30 seconds of poking buttons the sound engine came back to life and the patch was back to normal again. Not sure how clouds works at all, I just turned knobs till it sounded okay. The elevator door recording was a fun texture to retrigger and play around with, that plus whatever clouds is doing started making weird digital artifact noises, nice to add to the soup.

I went to the record page and guessed that things needed to be first enabled, then monitored, then recorded since that’s kinda how the patching worked with all the enable/disable stuff. It seemed to work, but I had no idea how to rename patches, files, or anything else to keep tabs on what is what. I was able to get the sound off the SD card onto my laptop, threw it through a compressor on ableton, and here’s my first sketch with the SSP. :slight_smile:

Stream SSP Sketch 01 by postsolarpunk | Listen online for free on SoundCloud

So far it’s doing everything it said it could do. I tried to get Linux set up on my laptop using the virtual machine but then there were some problems with me doing that, but eventually it seemed to work. My evidence for working is I was able to do a google search on a web browser on Linux, so I haven’t gotten to the dev stuff yet, too busy with grad school and staying sane.

Cheers y’all.


I’ve no idea how we’d go about doing that but I think it’d be a great idea.
The way the manual is now is very confusing especially to newcomers. For example a lot of the modules manual now is totally outdated so it’s telling you stuff which is not at all relevant since any updates, which is quite frankly really flipping confusing. I know that with updates the relevant information is put in with the update post but it’d be really nice to have a manual for each module all in one place that could be updated when necessary.

I’ve got a Sunday free and I finally got my case back from repairs yesterday, so I think I’ll tackle this and post some sort of forum info compilation.

Edit: alright I’ll start something but we’ll see how far we get no promises lol

Second edit: messing around for 4 hours and helping out another user, the situation seems to be we all know what midi is, we can all look up midi module on the SSP forum, but there’s not enough step by step walkthroughs for what buttons to push and how to set that up. Like we know what flour and eggs are, we know what a cake is, we know we gotta mix them together but we’re not sure how. I think writing up some step by step guides for the following ideas would help:

  1. power on, how to update firmware, load samples, load vsts, navigate 4 main menus and the grid, save preset, name preset, basic pre-patching stuff

  2. oscillator, vca, envelope, output example

  3. receive midi input to play the above example, tie midi cc values to other parameters like ENV release time

  4. set up internal recording, let’s record two or three things to generate some samples

  5. let’s load those samples we made into SAM and / or Granular and load up PMIX to tie everything together. Use the sequencer and LFOs to trigger SAM and Granular and modulate parameters.

  6. set up as audio interface and record SAM, Granular, midi osc synth patch, and PMIX full stereo mix patch out over… Is that 4 channels of audio into a DAW

I think a roadmap of something like that would cover the majority of use cases for people. I’ve written up 2 in another thread just now and I’ve done 4 and 5 before. I know there’s ample resources and knowledge scattered across the forum for all of these steps, I’ll try and consolidate everything with pictures of my screen as well.

Try and update the manual forum posts with links to relevant material on the forum as you find them. For example, if a module has been updated but the manual page hasn’t, link the firmware update post to the SSP manual post. I’ll start doing this too.


~Super low priority~

Working on that step by step guide, messing with audio interface functionalities.

Can someone tell me how to change the bit depth? It’s set to 32 bits, is there some way to change it to 16 or 24? I feel as though I’ve pressed every button I can think of.

I’m trying to debug glitches in the audio I’m hearing. Notes so far:
Windows 11 Ableton 11 Suite
MME/Direct works okay. Line (Percussa) Audio comes up. Only 2 inputs, 1&2 from Out1 and Out2 on percussa. I was hearing some clicks and realized my Out3 was sending a clock signal 0 to +5V out of it to an external module, but this driver thought it was part of the audio or something and I’d hear it click on the stereo input audio channel in Ableton. I only had 2 inputs when clicking [Input Config] instead of 24 as expected, but I think this is because the driver itself is limited.

I read that WASAPI is recommended by Bert, but Ableton doesn’t support it currently as of this writing. I downloaded ASIO4ALL v2, restarted Ableton, now it has a box [Hardware Config] that when I click it I get this box:

Which is cool. Realtek is my computer, HyperX is my headset, Percussa is SSP. I am now able to click [Input Config] and I get all 24 inputs as I was expecting. I am able to route the input from percussa to output hyperX. I’m getting some noise and I’m trying to debug it, but there’s several places where the sample rate and bit depth could be set that’s causing the noise?

  1. The sample itself: If the sampler has a lower sample rate or mismatched bit depth to the sample it’s playing, does this cause noise?
  2. Percussa audio at 48kHz, 32Bits
  3. HyperX audio at 8-48kHz, 16Bits
  4. Ableton Sample Rate 96kHz, ? Bits

Everything is set to 512 Buffer Size, but I’ve got no idea if that helps or not, gunna watch that vid in the Hybrid Systems SSP Manual forum page

this would all probably be better off in a different thread :wink:
… actually there are already a few threads discussing windows setup, so probably worth checking those out.

I do not believe you can set the bit depth… it is what it is. (32 bit is what you want anyway, as this is what its using internally, 16 bit is very low for todays standards, which are normally a min of 24 bit)

what I would be trying to do is make sure your systems is not doing any resampling, by ensuring everything is on the same native sample rate … so 48k
so, make sure you have the SSP set to use 48k for processing, put ableton on 48k

then you’ll need to play with buffer sizes, start with high numbers (even 1024) and don’t get any audio artefacts - then slowly reduce until notice audio artefacts… then increase again…oh and give yourself some headroom… to allow for processing in ableton (e.g. fx, vsts)

apart from that, id refer you to other threads on how to check where audio glitches are coming from.
people seem to have ‘different experiences’ with windows, which is partly the nature of the beast as windows hardware and drivers differ so much.

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Thanks again! Didn’t know if there was some problem between headphones at 16 and SSP at 32, just wanted to check that possibility. Probably should be its own thread lol didn’t want to start another one for a quick question like this. I’ll keep tinkering and snooping through forums thanks!

I suspect that Ableton will see the asio4all as 32 bit, then asio4all will downsample that, if its doing to a 16bit device.
the first thing Id also be testing is that Ableton is working fine with your HyperX thru asio4all… e.g with vsts.
basically check one thing at a time… before using combinations.
(then I guess check what options are for asio4all, and what they all do…)
it’s obviously very important that you know your audio system is very stable before you start adding the SSP to the mix, so that you don’t end up chasing your tail around in circles.

another option (for testing) might be to look for a daw (or whatever) that’s supports WASAPI, just to get some experience if ASIO4ALL is introducing issues (or not).

I know @MOTOKO uses Ableton CV tools with the SSP (as I’ve seen from their spectacular video series), so I know that Ableton can interface with it cleanly, but the rest would depend on Motoko’s computer setup too.

I’ll iterate through each combination sometime this week. Ableton to HyperX using asio4all is working well so far, just seemed to be a bit finicky with the SSP in the mix. I’m not too worried about it right now :+1:

Are they using windows?

I use ableton with cv tools too, but it’s on a Mac, which is very straightforward to setup, it doesn’t have these different apis/compatibility to contend with.
It’s not ableton, it’s rather the reliance on asio4all that windows introduces, that’s the question.

You are awesome for taking the time to do this in-depth response.
If it hasn’t been done yet someone should get you a cup of coffee.

As a side note I love your plugins for the SSP.

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