Total Noob Questions :)

Hello Forum! I’m brand new to the Percussa SSP- arrived yesterday- and I’m currently getting my brain around the inner workings. I’m sure I’ll have a million questions but here are a few that popped up right away- any help would be greatly appreciated!

  1. USB device port - not showing up in my mac either as audio sound card or midi device- is that normal? Are there drivers?
  2. USB host port 1 - doesn’t recognize a keystep pro or korg minilog, though USB host port 2 did, no problem- is that normal?
  3. In the manual, it says there are global DELAY and REVERB, but when I page over in my global menu there’s nothing there- is that normal?
  4. Can the track recorders record CV?

thanks in advance for any advice! oh and yes, its got the latest software on there - 28112020

Welcome aboard!

  1. Check the host software that you are using. No drivers required. Whne using Logic Pro, the SSP show up as 24-in 24-out in the in/output selection menus. No config required - true plug-and-play
  2. Only the middle USB port (2) is to be used to plug-in MIDI Controllers. This is normal
  3. The manual possibly refers to the very earliest firmware. A few iterations ago the global REV and DLY were removed. In place came to independent modules that can be used anywhere in a patch. Read up on the firmware update posts. They give you the latest and greatest descriptions of the new features.
  4. Yes. They record anything you throw at them. Pretty much across the SSP there is no distinction between CV and Audio signals, as in, you can patch anything into anything

great- thanks a lot, I appreciate the help! I’ve been grooving through various posts on the forum trying to get a feel for where the developers are at- I have a bit of that feeling the octatrack gave me… why am I always doing this wrong! ha. But I’ll get there. Still can’t connect modules without trying six times :slight_smile:

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ok- I’m tearing my hair a bit :frowning: More total noob questions:

  1. building a simple subtractive voice, do the envelopes need to control VCA’s, with the audio passing through a VCA? Or does the LFO module AMP act as a VCA (so you connect the envelope directly to the amp?) Either way, I’ve been trying to build a basic mono synth (input into LFO into Filter into Outs with envelope controlling amplitude) for three days and I cannot understand the module connection process. Its going Right-to-left, which is breaking my brain, and feels very easy to trip up because of needing to ‘enable’ ins and outs as well as make the connection. So confusing and I can’t make a single voice successfully connect to an output unless its direct (as in always on)- neither with Midi nor pitch/gate. I’m super duper lost.

I’ve seen on here that I’m not the only one who’s bumped into walls a bit trying to figure out the connection process- link, enable outs, connect, enable ins. Seems simple but right now I’m utterly blocked and can’t enjoy any sounds other than presets- major sad face. Plus I have no gauge on why ‘enabling’ is a part of the process at all. I’m not seeing any cool uses of enabling that could be accomplished with other ways of turning off/on signal paths. If its saves cpu to have them disabled unless needed, couldn’t connecting an input to an output also enable those ins and outs under the hood, so to speak?

I’m coming from ER-301 and Zoia, on which I was up and running in a few hours, so I just can’t figure out what the heck I’m doing wrong. Maybe I suddenly got dumb? But yeah, I can’t make a single voice.

Are there any videos anywhere on how to make a simple voice?

It seems like you have the patching concept right: 1. connect two modules 2. highlight (scroll) the output-input combo in the right hand column with the 4th encoder - click on the encoder to enable it (the sending module needs to be selected / highlighted in the grid) 3. select / highlight the receiving module in the grid and enable the corresponding input in the left column with the 3rd encoder. you will see the input counter increase from [0] to [1]

There are many ways to skin the cat of levels in the SSP. In your example it is best to view the LFO Amp input is an oscillator output amplitude control. The VCA is classic VCA with Signal input (A) and Control input ©. Oscillator/Filter output goes in A, envelope output goes in C.

BTW - Being the SSP and a modular system, in the context of a simple subtractive voice it would be perfectly fine to control the LFO output level directly with an envelope, but that would affect the signal level that feeds the filter so might yield slightly different results.

Anyway, to help you on your path, I have attached a Simple Subtractive patch for you to study. Unzip the attached file and copy the 060.pbp file to the Presets folder on the SD Card. (If you want the patch different location than 60, just rename the file to a different number first).

In this patch:
SSP Physical input 1 takes a CV signal for oscillator pitch
SSP Physical input 2 takes a gate signal to trigger the envelope
SSP Physical output 1+2 present the filter saw wave output signal.
In the patch the Envelope controls both LP Filter Frequency and VCA Volume

Please note: Some of the modules when inserted into the Grid have default parameter settings that are not always in-line with what you may intend. E.g. the VCA is half open by default, the ENV is looping etc. Don’t be afraid to dive in and experiment. Start with simple patch ideas. The SSP has its own way of doing things which is not always intuitive for everybody. As many on this forum have said, spend time and get your hands dirty.

Hope the attached patch helps you on your way.

060.pbp.zip (6.1 KB)

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your very kind, @titaanzink thanks for your patience and the help. I’d wanted to dissect the ob416 patch, but it was distorting, as were a bunch of others. I somewhere found the post about memory allocation + VST’s, so miraculously found an answer for why it was crumbling on me. I flashed two brand new SD’s today, making one with fewer of technobears genius plugs, but I guess even having any VST’s in there triggers the memory reallocation? Either way, I know have 3 SD cards but can’t hear the factory presets without distortion. I’ll go back in and remove all the VST’s from one so I can look inside those factory presets.

Thanks again. Trying to stay patient but this module connection thing is kind of a buzz killer for me. @bert Can you tell me why inputs and outputs of each module need to be enabled? I’m curious what the benefit of adding those steps (enable the outs, enable the ins) is? Genuinely curious- zero snark implied!! Seems like connecting two modules (for instance envelope out to SVF freq in) could also enable them, without adding the two extra steps, no?

And… I’m back :slight_smile: @titaanzink thanks again for the preset. I currently have the Percussa mounted in a waldorf kb 37 and using the pitch and gate, no problem- totally makes sense. My first task was to change the inputs from pitch/cv to midi. So I connect a midi input from the Waldorf (it shows up instantly on the percussa) and connected pitch to LFO pitch and gate to ENV gate and… it only plays every fifth or sixth note I play, and its playing them very very late. I’m transmitting on channel one, and I can’t find any midi input filters or submenu’s that might be related. In the midi input, when I focus on it, I don’t see any metering or midi input activity indicator, so I can’t tell what’s up. I tried two other keyboards- key step pro and org minilogue, but I’m getting the exact same effect. Thoughts? Ideas?

Not bert, but fellow noob. It was designed to that you could enable and disable inputs or outputs quickly, like if you clicked one button and disabled 5 connections, then hit it again and those 5 connections come back up again.

I’m pretty sure it’s like if you had 5 things connected to 1 input, you can disable and reanable that input and all your connections are there, instead of having to select each of the 5 connections and disable them all individually. This turned into a bit of a walkthrough as I studied how to answer your question, start with a blank grid

  1. Load ENV and VCA into grid, nothing in either’s right columns

  2. enable ENV to VCA connections, now both right columns have each other’s output targets:
    VCA can select to output to ENV ports
    ENV can select to output to VCA ports

  3. scroll to ENV
    scroll through outputs using 4th knob
    the outputs can be read as “output type : destination module : input port on destination”
    for example “Out:VCA:C1”, envelope out sent to VCA input C1 (C1 is gain, learned by reading VCA)
    click 4th knob to enable, should be lit up

  4. scroll to VCA
    scroll through inputs using 3rd knob
    the inputs can be read as “module : name of port : number of sources fed into this port”
    for example “VCA:C1 [0]”, VCA module, input port C1, 0 sources affecting this
    click 3rd knob to enable, should be lit up

  5. Once the output and inputs are enabled, you’ll see a blue arrow drawn from ENV to VCA
    ENV is now modulating C1
    On the VCA inputs, it should read as “VCA:C1 [1]” because there’s one source tied to this input

Okay that’s the basics, but to get to your question, what point is there to all the enabling and disabling. So let’s do something weird:

  1. Load up 2 more ENV modules, we should now have a VCA and 3 ENV modules

  2. Just like before, enable connections from each ENV module to the VCA module

  3. Enable ENV output to VCA C1 input for each module

  4. On VCA it should now read “VCA:C1 [3]”

This means we have 3 different ENV modules affecting a single VCA gain knob. In a modular environment, to mix CV signals you’d need a dedicated CV mixer, but I think here we can just keep piling up inputs. We’re using CV in this case, I’m not sure what would happen if we start piling up multiple audio inputs to the same input of the VCA. Let’s see:

  1. Load an OUT module

  2. Enable VCA to OUT connections

  3. Scroll through VCA output list. Notice how there’s a TON of options now. This is because we have enabled connections for the VCA module with 4 other modules. Again:
    the outputs can be read as “output type : destination module : input port on destination”
    for example “SUM:OUT:In1” means the SUM output of the VCA, fed to the OUT module, into the In1 port on the VCA
    Note, there’s OUT1, OUT2, OUT3, OUT4, and SUM types of outputs for the VCA

  4. Go to VCA. Enable “SUM:OUT:In1” this means that the sum of the audio fed into the VCA will be sent to the OUT module

  5. Go to OUT. Enable “OUT:In1 [0]” on the OUT module. Note how it should change from [0] to [1]

I guess we need a sound, let’s patch an LFO into the VCA audio input

  1. Load LFO

  2. Enable connection from LFO to VCA

  3. Go to LFO. Enable “Sn:VCA:A1” output on the LFO. The LFO has multiple output types, Sn, Sw, Tr, Sq, which are Sine, Sawtooth, Triangle, Square and other inverses of those shapes. We send this to VCA input port A1. We currently have those 3 ENV affecting C1 which is the gain for the first channel of the VCA

  4. Go to VCA. Watch your volume!! (If you have not done so, please use a patch cable to connect output 1 on the front panel to your favorite audio output source)
    Enable “VCA:A1 [0]” which should change from [0] to [1].
    After enabling, you should see the Sine wave from the LFO fed to the “VCA:A1 [1]” port.
    Scroll down to the output of the VCA the “SUM:OUT:In1” output port.
    You should see an attenuated version of the sine wave we have fed through the VCA to the output.

I turned down the Gain1 value to zero in the VCA module patch page. I turned down the values for the ADSR for ENV 1. I disabled connections for ENV 2 and ENV 3. Now it sounds like an oscillator rising and falling in volume.

Let’s say you had ENV 2 tied to C1, C2, C3, C4 on the VCA. Disabling connections from ENV2 to VCA will disable all the connections, but you can Enable it again and all the connections will be back again. Let’s say you had ENV2 outputting to 20 different modules. Instead of scrolling through miles and miles of inputs and outputs, you can quickly enable and disable patch points once they’re set up.

SO I PATCHED ENV2 to ENV3 and VCA. Then you can use the In/Out button on the front panel to see what is coming into the module or what is coming out of the module and it’ll change the blue lines around. Then I replaced ENV2 with a NOI noise generator module to see if things break, BUT the input and output connections remained the same while the module itself just changed into a noise source. I was then able to replace NOI with ENV2 and all my connections were still there. Interesting.

Okay so after typing all this I guess I learned:

  1. We have input mixing for each input per module
  2. We can enable disable module A to module B connections for quick manipulation of patch connections
  3. We can preserve the inputs, outputs, connections of a module on the grid while changing the type of module it is. For example, if we had an ENV modulating two other modules, ENV and a VCA, you can swap it to an LFO module, and the connections to ENV and VCA will be preserved.
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Thank you for this walk through- very helpful! This should be the updated quickstart guide!

After your explanation of enabling and disabling outputs and inputs (thanks for that) I would still suggest that having them enabled as default would save steps and confusion, but that’s just my opinion. If I were feeding 20 signals to one source, I’d use a PMIX or BUS just so I had a dedicated place to work with that many sources. Maybe a CVMIXER would be a useful tool, like a bus for modulations rather than audio.

On another note- @titaanzink I recreated the patch on a fresh preset and didn’t have any midi input problems, so I’m guessing there are midi input settings that came with the preset, perhaps? I saw something somewhere in a post that technofear had created a midi filter page, but I can’t find it.

Lastly, I still can’t get the device port to work at all- it doesn’t show up on my computer as a device for either audio or midi. Any suggestions on what could be happening there? I’m using a Mac, 10.15.

Thanks everyone for your knowledge and patience.

Thanks for reading! :slight_smile: lol

“inputs enabled by default” - yep that’s been requested a few times around the forum haha

“feeding 20 into one source use PMIX or BUS” - yep that’s what I’ve been doing too so I can control attenuation. There’s also a matrix mixer module that can do a CVMIXER type of job, where you can route/mix X amount of inputs to Y outputs. Does PMIX work as a CVMIXER? Might be worth it to test out what happens if you throw 2 LFOs into PMIX and see if you can route the CV through Aux or through the output. Not sure what the “optimal CPU solution” would be if you’re trying to mix and tweak a bunch of CVs at once, but I think there’s multiple solutions that exist so far. :slight_smile:

Oh duh, matrix mixer is exactly the thing. Boom.

hey- any thoughts on on those other two questions:

  1. is there a midi settings page I’m not seeing?
  2. why isn’t the percussa showing up as a device on my mac via the device port?

HUGE thx- Alex

Glad to help! I’ve learned a lot this afternoon myself! I’ll get back to you on midi for more detail walkthrough type post later because I need to step through it also trying to connect a deluge to the SSP. I’d start by reading through the posts about midi module on here, try and get just a single note on / note off thing going. Try and tie a midi note on to an ENV gate input. I know the first preset is set up for 8 voice wave table midi. Look through the documentation for Technobear’s midi modules, he’s created several for monitoring midi. The middle USB port is for midi device connection while the SSP is for host. I have still yet to test how to use it as an audio interface and connecting it to my windows PC / Ableton, I’ll be sure to come back with a walkthrough of that as well.

I think instead of recompiling all the manuals pages and stuff together, I think it might be useful to create a dedicated forum thread of step by step walkthroughs to get people practicing patching while also learning the quirks of the system. Guidebook tutorial handholding, since we know what midi is but we don’t know how to do SSP style midi lol

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it will appear as an audio device to the mac …
not usb midi device - this is not supported, the ssp act as a usb midi host (not device)

should work fine…
I assumed you are plugging the usb A (aka host) into mac into SSP usb B port (aka device port) then in a daw (or whatever) you should see the SSP as an audio device with 24 in/out.

Hello @thetechnobear - and thanks for all the amazing VST’s you’ve put together. Very rad. Ok interesting re: the midi, but nope it doesn’t show up as an audio device on either mac I have tried it on. Any other suggestions?

Also curious about computing power situation- I was trying to build a 4 voice synth, and its just crumbling on me. My understanding is this beast has a lot of power under the hood, but I guess with any VST’s going it really can’t handle 4 voices? Has that been your experience as well?

The voice I was building used the MMX4 as a VCA for each of the wave forms out of the LFO, so I guess its a bit more intense than a standard OSC+sub :slight_smile: What can I say, I’m greedy :slight_smile:

re: audio interface
there is nothing more to it that plugging it in… it appears as usb class compliant audio interface - are you sure have you got the latest SSP OS installed? not just the latest SYNTHOR, you need the newer image as well.
other than that, if you have connected correctly, I guess check the cable?
… can’t really think what else to suggest.

I’ve connected the SSP to multiple macs including the older intel and m1s on various versions of macOS, also to window and linux machines - and it’s appeared as an audio interface to them all for me.

4 voices has been fine in my experience…
bare in mind, I (obviously!) have all my vsts installed at all times on my SSP, and not had any issues…
that said, patches can be built more or less ‘efficiently’ , so hard to know if what you’re doing is too much or not, though with little details you mention, it doesn’t sound particularly cpu intensive to me.

as for cpu power, well its a quad core arm chip, so its no where near as powerful as (e.g) a desktop. but it is pretty powerful compared to what’s available in eurorack, which are often single core arm processors (like you er301) - though cores are not the only thing that drive performance :wink:

from your previous posts, it sounds like something is ‘misbehaving’ on your SSP.

what I would suggest is:
you ensure you’re running the latest OS image, and version of SYNTHOR.
make sure you have are running on the extreme sdcard that comes with the SSP.
initially don’t install any vsts, and check that the factory patches are running as expected without audio glitches, and other functions like audio interface are working as expected.

… this will give you a ‘clean’ base to work from, which you can then start building on, by adding vsts etc.

Hi @thetechnobear and @bert thanks for your help. I’m using the disk that came with the machine, and I’ve also followed the directions and flashed two disks that are brand new (32 gb sandisk extremes), and either with or without VST’s in the plug ins folder, the presets (such as 8vwto) cannot play- they’re just intermittent and distorted. Is it possible that this machine is broken in some way? I just got it used so I should figure that out if I need to return it or make a claim. Its very expensive, and I’m bummed that I can’t get it to function properly, but I’ve put in a lot of hours, and even starting from scratch I can’t get a 4 voice subtractive synth to work without it distorting and crumbling- clearly there’s something going on if that’s the case across 3 ssd cards, no? Any advice greatly appreciated!

no - it’s not normal for there to be distortion or glitches.

sorry, but Im getting totally confused about what problems you are having, since you keep moving between different topics - and you aren’t saying what’s going on very clearly.

a) the new OS image is required for USB audio interface to work.
and make sure you use the same sample rate on your computer and SSP.
use 48k sample rate on both.

b) try without any VSTs , just to eliminate them from the equation.

c) distortion…

first scale down signal levels, to make sure you are not clipping.

Id personally try with a sine wave, and then reduce its level, so you know its not clipping at output stage.

if you are not using PMIX (since you don’t have any vsts installed!) then you will need to ensure that the levels are a bit level, just in case there is a DC offset.

try first with main outputs , not usb…
remember that eurorack levels are ‘hot’, you should attenuate them when feeding into mixers/audio interfaces that expect line level.
… you can do this in ‘software’ within the SSP, simply scale down the output signal…

at this stage there should be no distortion/clipping…
you should check the scopes on the output module of the SSP to check for clipping (which is why a sine wave is useful, since its easy to see)

so now, you should know if the output stage is ok, and also there is no internal clipping.

if that’s ok, then you can try USB. … this should ‘just work’

if not, then you there could be a problem on your computer side, just like any audio interface, if you have rogue processes, or things like wifi, specs of the machines -these can cause buffer under-runs which are going to sound like audio glitches.
Ive personally not experienced this on the Macs that I use when I use 48k sample rate.
I cannot remember what buffer sizes I set, probably something like 256 samples, but it depends on specs of your Mac.

specific note on polyphonic voices:
of course, when playing multiple voices , each new voice is summed, so you have to compensate.
that’s just dsp for you :wink: , so if you 4 voices, you will potentially need to divide each voice by 4 , so its only 25% of full gain. in practice this will likely be a bit quiet, but that’s the theory.
note: this is not quite how polyphonic synths work, they will tend to use a kind of compressor… this means that when you play 1 voice its about the same as playing 4 voices.
(again this is not SSP specific this is just the nature of dsp)

whilst the above may seem like a lot, its not really…
basically it comes down to just make sure the gain structure is correct…
what concerns me, is most of this you have to deal with eurorack generally, so, I suspect you have already looked at.