#2028

Generally speaking, in my experience, simple attenuation is enough to get signals down to line level. I’d recommend bastl’s dude mixer — it’s not in rack and it doesn’t preserve stereo (it’s dual mono tho which can be good for getting mono euro signals to come out of both headphones, or for messing with feedback), but it handles all different levels really well, has mutes, it’s quite cheap/diy, and can use battery power. I’ve used it with all different set ups, great for mixing euro and non euro. Love my lil dude 2 death.

4 Likes
#2029

Depends a bit on the interface, but yeah. I’ve run straight out of several mixers to my interfaces (Presonus Firestudio Mobile and Babyface Pro) and also to Digitakt and Octatrack with no problems. I believe only XAOC Praga of my mixers was advertised as having line outs. I’ve found gainstaging inside the modular with different elements to be much more important than what the output module “officially” puts out.

2 Likes
#2030

I guess mimeophon will be a good match at it is a really good module. I have one paired with a Tiptop Z5000 / clouds / nebulae and this is offering a ton of sonic possibilities

By the way nebulae is a great module not only for granular but also for the different instruments mode you can load

1 Like
#2031

you can. Rings is in every rig out there (mine included), so don’t feel obliged to hold on to it. Be sure to explore the alternative modes (flashing LED), but ultimately if you find that the resonator part does not appeal to you, remove it. I personally like to explore the edges of its parameters - feeding it audio-rate stuff and momentary hi/lo signals as modulation. But it sounds like it sounds. It kind of needs to be sent into another module. Granular is an obvious choice, but e.g. a Mimeophone pairs quite well with Rings.

Rene v2 offers a lot more in the same form factor, but if you’re not into programming sequences, even on the fly, then that’s not-insignificant hp space that you can give to e.g. a Morphagene, Mimeophone, and Wogglebug…

1 Like
#2032

that’s good advice, thank you. I do agree and also enjoy pushing certain modules, and most of gear actually, to it’s limits with audio rate signals, etc. the alternate mode is ok. and since learning of the lofi/delay mode here with the negative fm voltage, i was pleasantly surprised. but i still feel like it’s not a very strong or flexible sound design tool. i may be wrong, i still like it, it does a few things very well. but i am really asking what type of modules exist that would sort of equate to the type of sound sculpting utilities or experimental processing tools available in something like ableton. a resonator would be one of those tools. i just don’t think that’s one i can make much use of. i think my idea of modular has been warped by the unrealistic flexibility of vcv rack (like running audio through a bernoulli gate to create particulate of the external audio). so these modules may not exist, but the idea that i have in my head is having a rack full of modules that i can take an external signal, run it through all of them and mix signals, layer them, splice them together, use subtractive tools to carve out spaces and break them down, effects to alter create new environments and spread pieces of sounds across those spaces. so i’m wondering if there are maybe some out there that i’m not aware of. some of the xaoc stuff seems like it would be good for sound sculpting. some of the industrial electronics modules, make noise, modcan, instruo. i haven’t used any because i’m new to modular but the goal is to create in hardware what most other hardware gear cannot do as far as sound design.

i try to study people like rashad becker, valerio tricoli and jan jelinek. a lot of the grm artists and other concrete artists. it seems like they use reel to reel tape a lot. but they also use modular. i may just have a problem where i cant accept that a daw is the only way to achieve what i want and what im used to, but if there are some pretty innovative sound dismantling modules, i would like to replace what i have with those

#2033

Layering and mangling sounds is exactly what I do with eurorack. I can’t speak for the results but the process to me looks like what you are describing.
From my observations sample based modules, like Morphagene, Lubadh or Nebulae and multi input/output filters are best for this, at least in terms of speed and flexibility. Essentially splitting/cross-fading audio into different processing chains and then reassembling them.
I too struggle to find use for synthesis modules, as I find myself unable to produce quality results a lot of the time. Often times the sound is really stale or very chaotic.
There is one thing I gather from your posts about eurorack. It’s the desire for it to be quick and immediate in its results. My personal reality is that its immediate in its process and unpredictable in results. As the complexity of the patch grows it gets harder to switch things fast, and at some point it’s easiest to start anew.
Also I found it really hard for myself to interact with a lot of devices at the same time. Since getting octatrack I used it with modular a couple of times, but most of the time both of them are just too much for me :slight_smile:

3 Likes
#2034

i absolutely love nebulae. its a dream module. i could be happy with that for a while, but the results i get from that dont feel like i really constructed them. there’s nothing wrong with that. it may be the best possible tool for this,but i would like to add some more functions to my little setup. i have this like mental problem where i want to make everything from scratch or else i feel its worthless. which i know is ridiculous and anything i do make from the ground up sounds like some 14 year old’s asmr youtube video. probably worse. but i can’t get past it usually.

i was looking at qpas. those sorts of things with unique functions are interesting. i imagine a system where i can take a raw basic noise source like short field recording, noise burst, oscillation; run that through filter, wavefolder, some kind of signal blender (modemix?), downsampling, cross modulation, frequency shifting, saturation, some module im not aware of and come out with something new. it would be cool to have a bunch of basic tools that have a lot of fine tuning ability. like basic function, but more options for tweaking.

i was praising the digitone in the polysynth thread but if i were to criticize it, i get really frustrated that it does not have more options to take a raw basic sound and route that through many individual functions that i can carefully chip away at and fine tune. it’s close, and elektron does a good job with combining flexibility and ease of use. but i wish i had more options to further create my own sound from the source material. i just want like a little chemistry set for audio. might be unrealistic. i’ll work on it.

maybe that koma field effects kit would be good

#2035

I think best tools are shortcuts to what makes us “click”. Your words sound like you search for the exact opposite of “shortcut”, while also wanting for it to be compact or contained:) I honestly think that you need either a big case of modules or a nice way to integrate pc into your setup.

2 Likes
#2036

So I’ve been using an ES-8 for a bit now, and it’s great for so much of what I do. The thing is, I’m always, always, wanting more outputs. My heart paused a beat when I saw the ES-9 thinking that it had 14 outputs, or when reading the (“16-in/16-out”) in the description only to be disappointed to see that it was still only 8 outputs, with 14 inputs.

I’ve looked the Expert-Sleepers page a bit and beyond find all the expansion names/functions confusion, think that nothing will let me expand the ES-8 if I’m using the onboard ADAT I/O to connect it to my audio interface already. (Happy to be corrected here if I’ve misread that (actually, can you use an ES-9 + ES-5 + ESX-8CV to have 16 CV outs + 8 trigger/gate outs?))

So at the moment I’m looking to see what the options are for having “loads” of CV outputs coming from my computer. Ideally this would be at audiorate, but curious to know what other options are available. I’ve seen a bunch of MIDI-to-CV things, but wondering if there’s anything in the i2c/crow landscape that would do allow for something similar. Like a crow + some expanders (or multiple crows) or something. From the crow page it’s 16bit which is great, though doesn’t mention the CV rate.

#2037

You can use 6 of the expansions ports of the ES5 for a large number of CV/gate/midi outputs

I use ES5, to connect ESX-4CV AND 3X ESX-8MD = 30 plus outputs, also use ES6 and ES7 for inputs - endless possibilities!!!

OS is on here Rodrigo and is most helpful

couldn’t live without expert sleepers :slight_smile:

#2038

So I guess an ES-5 can connect to an ES-8, but then I would lose two of my outputs from the ES-8?

The ES-5 itself is just binary outputs, so that’s handy for that kind of stuff, but with additional ESX-8CVs, would that then gobble up additional “real” outputs from the ES-8 in order to add more CV-rate outputs from ESX-8CVs?

So, if I had an ES-8 + ES-5, I could then have, um, 4 ESX-8CVs in order to have 32 low-res-ish outputs with no functional outputs from the ES-8?

And if you go from an ES-9 does the piggybacking of an ES-5 take one of the minijack outputs, or can it absorb from the headphones or wherever else?

#2039

So this “make everything myself” thing I TOTALLY get! I have a similar sort of… “false image” of what it means to create that I tend to hold myself to.

The issue is that I’m not Isaac Newton, Tesla, Don Buchla, and Bill Evans all wrapped into one. Gotta stand on the shoulders of giants and use whatever tools you have. The other anecdote here that helps assuage that feeling is: a painting is done when you can stand back and say “I did that.” Spoiler: that never happens.

I’ve nourished a focus on “what moves me?” over “what can I do?” – it’s not about me and my abilities and my expression, but about the connection between my ears, hands, and heart, and about casting a moment I perceive as beautiful in the best light.

It’s sometimes hard to maintain this mindset, but it’s MUCH better for learning, and for having the confidence to press record – it’s just the most beautiful thing you saw today, not the most beautiful thing you’re capable of rendering. Always witnessing.

I know this is sort of an abstract, mind set thing that isn’t really what you asked, but I identify with the attitude you expressed and realized, over years and years, that it was hurting me.

6 Likes
#2040

I think it makes more sense when you learn what is actually happening with the expanders.

Each individual output on the ES-3, ES-8 or ES-9 are 24-bits, running at audio rate (they are just DC coupled audio outputs).

The ES-5 takes two of those 24-bit channels, and allows you to repurpose the total 48-bits for other data. Specifically, it allows for the connection of up to 6 expansion modules (though the initial ES-5 counts as 1 of the 6, if you want to use the ES-5 outputs). Each of those expansion modules uses 8-bits of data (24-bit stereo pair x 2 = 48-bits, 48-bits / 6 expanders = 8-bits per expander). For the gate output modules like the ES-5 itself, or the ESX-8GT, the 8-bits are used to sent gates / triggers at single sample precision (audio rate). For the CV output expander, the ESX-8CV, each 8-bit sample is instead time multiplexed across the 8 CV outputs, which are each 12-bits but individually run at a fraction of the audio rate of the project.

Another thing to keep in mind, at least in my experience using Ableton Live, is that while it is stated that the ES-5 uses two channels of the ES-3 / ES-8 / ES-9, if you are only using <= 3 expanders including the ES-5, you only have to sacrifice one output. This is because the first of the output pair is used to encode the first 3 expanders, and the second of the output pairs is used to encode the final 3 expanders.

But for example, if you give up two outputs on your ES-8, you could gain 8-triggers / gates at audio rate (the ES-5 outputs) as well as 40 12-bit CV outputs running at a fraction of audio rate (5 ESX-8CVs attached as expanders 2-6 to the ES-5, each with 8 CV outputs).

2 Likes
#2041

Wow, thanks for the super detailed response! Os pointed to a useful thread on muffs that breaks some of those same points down.

One of the main points that I didn’t understand was the 24bit->8bit part of the process. Which makes what happens with the expands more understandable.

Does this 24/8 thing sitll hold true if you use an ES-8 in conjunction with another audio interface, and happen to be running at 16bit? Does it just 16/8 instead in that circumstance or does it bit reduce all the existing architecture?

Lastly, if I’m understanding you correctly, if I have an ES-8 + ES-5 + (1x) ESX-CV8 I would have:
7x audio rate CVs
8x audio rate gates
8x (1/8th (5.5kHz @ 44.1k)) 12-bit (4096) audio rate CVs

#2042

I’m not sure I’ve got this right, but I’ve been thinking that envelope clipping is what some envelope generators (e.g., Quadigy) simulate as “punch”, and the resulting effect on volume on a module that responds to less than 10v might be why a 10v envelope generator like Maths sometimes stands out?

2020-10-12 at 10.02 AM
2020-10-12 at 10.02 AM1260×728 36.3 KB

Envelopes are of same duration and ‘exponentiality’. In addition to the extended duration at maximum extent where the clipping occurs, note the difference in volume (solid green) between the envelopes.

9 Likes
#2043

That’s a good question about running at 16-bit. I personally use an ES-3 as an ADAT expander from my main interface, it sounds like you use the ES-8 in a similar way, rather than running from USB. I think the ADAT connection might always be 24-bits? I’m not sure how the DAW handles that if your main interface is at 16-bits. I also see that Expert Sleepers lists the ES-8 output as 24-bits, can you run the ES-8 at 16-bits if using as your only interface directly over USB?

On your setup, what you described is basically the case, except that the ESX-8CV sample rate is adaptive depending on how many outputs you are using. From the Expert Sleepers website:

In order to squeeze eight 12 bit channels through one 8 bit connection, the channels are time-multiplexed and so effectively run at between one-third and one-twenty fourth of the audio sample rate. For example, if the audio interface/DAW is running at 48kHz, the ESX-8CV’s outputs are running at between 16kHz and 2kHz, depending on how many channels are active.

I should also note that I use the ES-5 with Ableton while only consuming one output from the ES-3, however I’m not sure how that works on other DAWs (Ableton lets you route two tracks to the same physical output where I assume they are mixed)

1 Like
#2044

For my purposes, I’d only be using two total expanders, so the 16-bit would cover it one way or another. I just wonder how that works, because even if the ADAT connection is 24-bit, presumably you still have to pump those bits out from your computer (firewire/USB).

To be fair, I only ever use it ES-8 when I’m in Max, so it could be that I’m in 24-bits anyways.

Yeah, that’s what I was trying to make sense of. So presuming I’m running all 8 outputs on the ESX-8CV, would it be 44.1k (in my case) divided by however many outputs? (so in my case, the 5.5kHz rate)

What’s a bit confusing from the description is that it says between 16kHz and 2kHz, where 16kHz is 1/3 of 48k, so not sure how that factors into things.

So I guess if I have all 8 running, it would be at 2kHz each?

edit:
And I guess because of the time-multiplexing, all 8 channels of CV from the ESX-8CV would happen across a period of (in the case of using all 8 channels at once) 4ms?

edit edit:
As confirmed by Os on mw:

Lastly, if using an ES-8 + ES-5 + ESX-8CV, as I presently understand it I would have (presuming all 8 outputs going on the 8CV):
7x audio rate CVs
8x audio rate gates
8x 2kHz 12-bit audio rate CVs (time-multiplexed/staggered)

I guess the 2kHz would be slightly slower if I’m running at 44.1k, so 1.84kHz.

And @DMR, you were correct about the 24-bit ADAT thing happening independent of the DAW/computer settings.

#2045

you’re right there. i think i’ve concluded what i expected all along, and the reason i didnt even want a modular setup in the first place. which is that, for what im looking for, modular would only make sense if i had the money to build a huge setup with all of the tools i need. and you’re right about the pc too. if i can get over the idea that ableton is a detreiment to my music making, id probably be a lot happier.

the only problem there is that the only removable ram slot on my laptop’s motherboard is broken and im stuck with 4g. so that kind of sucks. tried replacing the motherboard even and it didnt work. so no idea whats going on there, but thats a story for another time. ableton is great.

@yarns i feel ya. i do try and keep that in the back of my head. i do also think that maybe my favorite part of music creation is sound design though. so i am hoping that someone like elektron pioneers some sort of all in one groovebox/synthesis & sample machine that has a bunch of experimental methods for meticulously crafting each little sound and structuring them on a timeline or placing them about organically in a virtual space. so its not always just about “did i do it myself” because i do realize thats unrealistic and a waste of time. but thats a very good point you make

on a more related note, the Lion module from instruo looks very interesting. i kind of dont want to give up the ability to at least try out the modules that do look interesting and arent really available elsewhere. its a weird thin line between going all in on a big modular setup, staying away from modular altogether, and deciding to go with a small one that you have to constantly change for using a small amount of unique sound tools

#2046

I’ve looked at this like three times and it should have been obvious:

that’s the same way the amplitude would be impacted by compression.

#2047

I don’t understand how. If you somehow applied compression to the envelope itself, I would think that would simply attenuate the envelope (scale it down) to some degree relative to its original amplitude. If you’re compressing the audio somewhere downstream, I would think that would merely attenuate the audio signal in a relative way, not reproduce the timbral- and relative amplitude effects baked in by an envelope “clipping” (the punch) at the CV input of the VCA or filter or whatever.