Noise module random bursts & general Audio output skipped every second repeat in same setup

Flavelius

Hi, this is a little hard to describe, but I put a high resonance bandpass filter after the noise module to get a smooth sine-like sound. But this sound seems to randomly get louder in short bursts. These bursts are not related to the note input. And they result in volume clipping.

In the same setup I noticed that when I increase the voice count and connect the noise poly socket that every second repeat of the pattern results in silent output even though a signal monitor on the last module‘s signal socket shows that there should be a signal (Amp ADSR is the last module).

Now I’m not sure if these are issues with my setup, my expectations or actual misbehaviours

https://streamable.com/xd9u1f

ScreencapDrambo.zip

rs2000

I haven't spent much time with it but my in first rebuild of your patch, I took the Limiter out of the polyphonic signal (you can always place it after the ADSR of course).

No sound issues so far.

bcrichards

Can't speak to the clipping, but noise through a resonating filter is going to be, well, noisy.

The skipping is almost certainly the AUV3 that you have in between the filter and the ADSR. Everything is summed to one single input going in to any AUV3 processor and it comes out monophonic as well. So you have four voices going into a monophonic input. I don't know for certain if the voices are summed or if only one gets passed through - then the ADSR amp is looking for four voices but only receiving one, so it only plays on every fourth gate it receives.

Flavelius

Placing the limiter auv3 after the adsr indeed fixes the skipping. I don’t quite understand the role of voices, so I didn’t expect that to make a difference. I only tested increasing the voice count in an attempt to make polyphony work which it kind of did, and now even more functional. Is there a way to convert the multi-voice input into a format compatible with auv3s so I don’t have to stack them at the end of the chain (which is not possible some scenarios) or to enable polyphony without using multiple voices?

edit: I just found a YouTube tutorial about voices and signals by I guess @bcrichards (thanks for these useful videos!), I‘ll likely find my answer there.

As for the noise, I thought the characteristic of white or pink noise is something like an equal weighted distribution, which in my assumption would result in a still noisy/soft output even from a small cutout section of the whole spectrum . I used majken‘s chimera vst on pc in the past and liked it so much that I wanted to attempt to rebuild some sounds in Drambo. It uses noise and resonant bandpass filters to shape its sounds but doesn‘t have these irregular volume swells/peaks. Maybe my approach is wrong or the bandpass too shallow?

rs2000

https://forum.beepstreet.com/discussion/comment/52247#Comment_52247

I'd really like to replicate your issue. Can you record a short audio file without the AUPeakLimiter and post the zipped file of your Drambo project and the audio file here?

Flavelius

https://www.forum.beepstreet.com/discussion/comment/52251#Comment_52251

Following your request I removed the peak limiter which returned the clipping; to not record it I decreased the output volume and - the issue disappeared.. I think I now understand that the fluctuations are a result of the reaction time setup of the limiter, similar to a compressor (in any case the ios provided limiter auv3 is the reason; unfortunately I couldn‘t find a Drambo replacement module). Thanks for leading me to finding the cause.

Flavelius

I also watched the tutorial about signals and voices, and while it gave me a better overview, i still have open questions. How do I know which modules or auv3s can take multiple voices, which only a monophonic signal, which stereo and is there a way to combine voices into the kind of signal a module or auv3 expects without nesting whole parts of a chain in an instrument rack for example (that the video showed does auto-combine voices as output)?

And why do modules that can accept multiple voices and are placed behind ones that don’t not automatically switch into single voice mode? Shouldn‘t this info about module’s capabilities be available to the chain?

Maybe I’m still misunderstanding a core concept here. In that case I’d be thankful for clarification.

(How is a monophonic and a stereo signal related to voices and inputs, are voices like serial batches of processed data for each module (unrelated to whether a signal is mono or stereo even?) ? This might explain the skipping in the entry question atleast).

edit: I found this section on the Drambo docs website:

„All signals in Drambo are polyphonic and stereo by nature.

Polyphony and stereo information is propagated through connected modules. e.g. a Module that gets a polyphonic/stereo signal works polyphonically and outputs the same signal type. 

TIP 

In order to save CPU, if you don't need a polyphonic signal throughout your whole patch, use the Poly to mono module. Some modules marked with (1) do this by default (e.g. Reverb).“

So, (1) is an indicator for modules that ˋflattenˋ the signal. Signals are always polyphonic by default. Does this mean the entry issue stems from a module not correctly flattening the signal itself and instead only ˋacceptingˋ a flattened signal (which according to my understanding of the above description shouldn’t exist, and using a poly to mono module anywhere in the chain doesn’t fix it, surprisingly)? It also means monophonic signals don’t really exist and are atleast stereo and it’s just the wording that’s confusing as it relates to voices instead of the conventional mono/stereo (?) I’m even more confused, sorry.

rs2000

OK I see, polyphony as it's done here is a Drambo invention, aimed at making polyphonic patching as easy as possible. And that works as long as you keep a certain order of modules and only use Drambo modules.

Let me repeat a few facts about polyphony.

• You can always visualize polyphonic signals by tap-holding a signal port and choosing Input or Output, then one of the little square boxes to select the voice number
• Polyphony usually originates from the MIDI to CV module from which you get Gate, Velocity and Pitch signals that carry as many voices as you adjust in MIDI to CV
• Using modules with (1) after the name or external plugins like AUv3 will always downmix the signal polyphony to 1 before feeding it into the module or plugin. From that point, the polyphony information is lost (but still available from before the downmixing module of course 😊)
• Amp Envelopes are quite powerful because they can combine the polyphonic Gate and audio signals by applying the adjusted envelope to each audio voice separately, triggered by the respective gates in the polyphonic gate signal.
• If you give the Amp Envelope a polyphonic gate and a monophonic audio signal when using polyphony N, only every Nth note will sound because only one of the voices is available in the audio signal
• If you give the Amp Envelope a monophonic gate and a polyphonic audio signal, all notes will always sound simultaneously because the Amp Env needs to downmix the audio signal to 1 voice as the gate signal polyphony has the highest priority in Drambo's poly world.

I hope that clears it up.

pedro

I just have to say I love these morning tutorials! Thanks @rs2000 for being such a great teacher and taking the time lay it out so clearly

rs2000

Thank you @pedro 😊

Flavelius

Thanks, that was very helpful information!

• Using modules with (1) [...]

As some setups with auv3s seem to misbehave in this case, is there a way to discard this polyphony information, so that later modules after such a downmix work correctly (not expect polyphonic input) or is this a case for a nested instrument rack?