Tonverk: Tips & Tricks

The thing about timestretch (at least for me) that makes repitching to change sample length not a good solution is that I want to be able to drop bpm temporarily as an effect and have it be like slowing the whole tape head, not just fitting a sample into a different bpm than I recorded it.

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I’m curious how other Tonverk users are dealing with levels. I noticed, that when sampling into Tonverk, at a level just below clipping the meters in Tonverk, the actual level of the audio is very high. If I load a sample recorded in Tonverk into my DAW, It appears to be high above clipping, but, since it is 32 bit float, it isn’t actually clipping, because you can just turn the gain of the sample down, and the sample isn’t clipped. I noticed that it ususally ranges from about 12 to 25 db that I should turn down a sample recorded in Tonverk in my DAW, to get it to not be clipping. So, because of that, if you import in files that are not 32bit float, with high gain, they will sound quiet compared ro internally recorded sounds. I put together a quick batch processor that will convert convert audio files to 32bit float 48k, and lets you add gain. So, I have been doing that to files that I import. It made sense to me, to match the level of files that Tonverk records internally. But, the odd thing, is that, when looking at the audio files that come in the factory library, they are not at high levels. They are all at or below 0. So, it makes me wonder if it makes more sense to turn down internally recorded samples to match the levels of standard not float samples, or if it makes sense to batch process everything to match the high levels of Tonverks’s internally recorded samples. This also goes for the multisamples. When you record multisamples, even if they aren’t clipping, they are far above 0, when you look at the wave in a computer. I made a little program that allows you to edit the multisamples recorded in Tonverk. It analyzes the level, and turns it down to non clipping, and exports an SFZ with a folder of samples, as opposed to the single sample you get with the autosampler. Then I can edit the multisample. The converter then allows me to add the gain back to the sfz, when loading it back into Tonverk, so the level is the same as it was in the autosampler. I’m going to share the converter on here when it is done. But anyway, I’m cuious how everyone else is dealing with the levels. Is it better to use the levels if its internal recorder as a standard, or to use normal levels, like the ones in its factory library as a standard, and turn all internally recorded sounds down. For now, I am going with high levels, and batch processing all imported audio to 32bit float, and adding gain. I’m also a bit unsure how much gain to add. I’ve been testing it, by recording a lot of stuff, right below the level of clipping in Tonverk, but then when I look at those audio files, the amount above 0 that they are ranges from around 12 to 25 db. So, I’m a bit unsure what is a good default amount for batch processing gain.

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I’m sure this is just simple maths but I’m having a lot of trouble triggering accurate fades using the LFO and amp envelope

the pattern itself has trigs on almost every step and I’m looking to fade it in slowly and smoothly over the length of one 256 step pattern - what’s the logic for this?

I’m only able to really get the fade to actually sound by setting the LFO to FREE or HOLD modes and using those places the LFO wherever it chooses, I thought the ONE mode would be the right choice here but that seems to trigger nothing at all? (global pattern setting and placing on the first trig don’t seem to make any difference)

I have tried just recording the fade using LIVE REC which will not work because it cycles through the trigs quicker than I turn the dial one value (so only catches a parameter lock for intermittent steps) - also have just painstakingly locked each step across even values from 0-127 but this sounds a bit blocky

anyone understand the process to just generate a smooth fade in triggered at the start of a pattern to reach max value on the last step?

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I did this empirically one time and I believe I ended up with the ramp LFO on half cycle mode with something like speed 2.0 and multiplier x2, though I had a shorter pattern than 256 steps and am not sure if the LFO is able to do 16 bar periods or if it is limited to 8 bars like on previous products.

What I do remember is the speed and multiplier settings that worked ended up different than I had expected from the LFO table in the manual. I also had to play a trick with step 2 trigger and microtiming to get the fade in to happen only on the first loop through the pattern, so the first step had the p-locked fade in parameters and only happened once and the other step was micro timed to be step 1 in the time domain and had different LFO parameters and happened if the previous step didn’t.

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LFO math over 128 steps to use as fader;
Eg mod destination send delay
Mod range 0-127

Most important- If using polyphonic mode, set PLAY MODE to POLY M.LFO (FUNC +TRIG)

LFO shape triangle
Speed: 4.00
BPM: 1
LFO mode to free

Plock step 1 & step 2 to retrig LFO (quite absurd that the device has broken retrigs but whatever. Step 2 retrig is to make sure the lfo 100% retrigs. Sometimes with one retrig it doesn’t work, another way is if u have 2 consecutive patterns with the same instrument you can retrig LFO on the last step of old pattern and first step of new pattern, which will also result in 2 retrigs ensuring the LFO restarts) or as jeffroman said hide the retrigs in microtimed trigs.

With these settings you will 0-127 or 127-0 over 128 steps depending on which side u set the depth i.e the triangle goes from neutral to max over this period, so if u want delay send to go from 0 to full delay send should be set at 0 and LFO depth to positive max. Inversely if u want delay send to go full to 0 set delay send to max and lfo depth to negative max.
For 256 steps you would need to lower the speed to 2.00.

This is applicable to the triangle/sin shape only. Now if u wanna use a saw shape I would suggest changing the start phase of the triangle to 90 (downward saw) or 270 (upward saw)

You could use the saw shape itself but that would change the math since the shape is half a triangle.

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I am not used to 32 bit float so I won’t talk about it, but what confuses me the most is that waveforms display is normalized, even if the sample isn’t.

Once I recorded inputs without nothing plugged intentionally : “silence” with maxed amplitude displayed…:zonked:

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hey this worked perfectly - seemed that the biggest issue was that I indeed did not have the trig settings to POLY M.LFO

just wanted to add some observations which I THINK is how this has worked in the end:

Speed: 4.00 + BPM: 1 = 256 step pattern (as suggested)

I used the ONE trig mode though and the RAMP wave and had no issues

I personally did not need to set the retrig or 2nd step in any way and the LFO was used on the global pattern settings rather than locking to the first trig

I BELIEVE the depth control corresponds to the amount that the LFO increases the value, for example - I wanted to hit my target max AMP volume of 56. Setting the depth to 56 and the global amp settings to 0 achieved this but the fade was too slow and the lower values were inaudible. So I changed the global AMP volume to 20 (the lowest audible level for my sample / mix page settings) and set the LFO depth to 36 - it seems that this started the fade where I needed it and it reached it’s peak before jumping to the next pattern

Haven’t tried yet but I’m assuming reversing this and ramping down will work the same for a fade out

Happy if anyone wants to correct any of this but this seems to have worked for me - thanks for all the help :slight_smile:

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Glad it worked out!
Thanks for the pointer, I shall too try with ONE instead of retrigging the LFO. Will save me a lot of hassle, I just hope it holds up through pattern changes.

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just to add some salt to my celebratory post - I tried it again on another mutlitrack pattern and it just doesn’t work so I’ve no idea why they are different or if it is just more funny behaviour

additionally I noticed the LFO on the track that worked (sometimes) behaving differently when using song mode so another mystery there

at least inching closer to figuring out how to properly work this thing :slight_smile:

Just in case anyone else was missing the speed of auditioning presets as much as me, I have noticed a weird trick. If you just press the Preset button and load a preset from there it will load it and close the preset browser so you have to keep reopening it, BUT if you press FUNC + Preset go down to presets, select Load you can then load a preset and the browser stays open so you audition presets quickly, I hope that made sense, its late.

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As there is not a ‘user’ or ‘fave’ tag for presets, I noticed that there are only 4 of the factory presets tagged with ‘epic’. I deleted this tag from the 4 presets & tagged my own presets with ‘epic’ allowing me to filter my own presets easily.

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Yeah it would be nice to have custom tags. I like to know which machine a preset uses. I have to include a W for Wavefinder or G for Grainer etc… in the preset names

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This doesn’t work for me on firmware 1.4?

Edit: ah, you have to go to settings, then hold track, arp and mute - and then while holding them, press “fill”.

not sure if this is the right place to ask…but is there a way/programm to check in what Project/pattern a specific sample is used?

https://www.patreon.com/cw/ArturPasco

Filter folder is ******* amazing!
It’s been my staple on fx slot2 and still remains after months of usage.

IMPORTANT: set IN to -32 & filter to OFF

At this setting your sound will pass clean through it with no change to amp or spectra. Check via bypassing it.

The IN values after -32 upto max are a distortion (I think, sounds like it) before the FOLD and DIST. Combinations of this can result in some super gnarly stuff and if u chasing loudness this can help a lot too.

The OUT governs the resultant output and I use it as gain staging, as for the filters- super handy for sculpting the sound but its important you are aware of the internal signal chain of the FX the get the most out of them.

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Thanks! IN to -32 changes the effect completely! Will use it from now on :slight_smile:

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My favorite as well. Works well with the Degrader on first slot ^^

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I often have the input at -35 or as low as -45 with filter folder, it lets you get a lot more useful range from the wave folder in my opinion. It’s my favorite distortion effect on Tonverk for sure.

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