some thoughts/suggestions

[brok landers]

hi chris,

i hope you're open for suggestions...

at first: as i see it you reuse voices. i.e. retriggering the same note uses the same voice. while this is great for cpu saving, it has one flaw that i've yet to be able to overcome (maybe i'm doing something wrong). when using velocity on the cutoff of a multifilter, the velocity stays the same as it was on the first note on, when retriggering the same note while the same voice still is in amp release stage. so, is there a way to turn off the voice reusing (of course at the expense of more cpu usage)? or how do i go about that issue?

i didn't find any way to control adsfr parameters with whatsoever input. i'd like to be able to control the parameters i.e. with cc's, velocity, or with lfo's or even other env modules. again, i might be doing something wrong.

when using an adsfr on the multifilter's cutoff, i wasn't able to modulate the ammount with whatsoever cc or velocity. is there a possibility to connect a source to an ammount, as soon as one connected a source previously? i.e. as soon as i connected an adsfr to the cutoff, i assumed to see this connection-ammount to be availlable in the multifilter's destination menue...

i see no way to connect a multifilter right behind the voice mixer. i wanted the multifilter to be monophonic, using a static highpass, as the pulsewave produces a _lot_ of bass. in order to remove the hefty bass content i only need a monophonic filter. is there a way?

you have a "constant" source availlable in the modulators. could you maybe add another one called "alternate"? it would just send 2 values, -1 and 1. every note on would switch beween those 2 values. when i.e. connected to the voice panning, one could spread the voices evenly in the panorama, as each keystroke would pan each voice oppositely. that would be great.

while we're at the sources, could you maybe add a "random" source to the modulators? it would send out one random value per keystroke. i know i could abuse an lfo for that, but for a simple radomizing per note on this probably would be more efficient than using an lfo.

the "pitch" (keyfollow) in the cc module has a e3(?) as fixed base note, which makes it impossible to use it on the spread of the supersaw. it's basically needed to detune more on lower note, but less on higher notes (sounds more natural). without beling able to specify the base note, when using the "pitch" as a source for the supersaw spread, the spread is hax (specified ammount value) at the lowest as well as at the highest note, getting narrow in the middle, where c3 does null it. so i vote for being able to freely adjust the base note when choosing the "pitch" parameter as a source. there's hundreds of other cases where this would be useful.

the keytracking in the multifilter is unipolar upwards only. i suggest you to make it bipolar, so that one can have negative values, the higher a note is played. i know i could use "pitch" as a source, but why waste a module, if the parameter essentially only has to be changed to bipolar? also here, an adjustable base note would be really helpful.

i would love to see a bypass switch on modules. sometimes i'd want to i.e. bypass the filter, or the extra osc that discretely modulated the cutoff at audiorate, i wouldn't want to be forced to always have to set the ammunt to 0.

why different modules for the osc's? i'd rather put a dropdown menue entry for the different osc modes. that way i don't have to delete > instanciate > recable every time, just because i choose to use another osc type.

that's it for now.
i have to say it again, in case all my "moaning" leaves a different picture - this is highly addictive. i'm really not impressed easily nowadays, due to the nature of my job, but thisone is a bright light, really. love it!

[sonigen]

i hope you're open for suggestions...

Yes!

at first: as i see it you reuse voices. i.e. retriggering the same note uses the same voice. while this is great for cpu saving, it has one flaw that i've yet to be able to overcome (maybe i'm doing something wrong). when using velocity on the cutoff of a multifilter, the velocity stays the same as it was on the first note on, when retriggering the same note while the same voice still is in amp release stage. so, is there a way to turn off the voice reusing (of course at the expense of more cpu usage)? or how do i go about that issue?

Is it that you specifically want to not reuse voices, or just that you want the fresh velocity to appear on retriggered notes? The former situation was designed to be like that because I dont see how you can reliably know which noteoff goes which which noteon if notes overlap. The later situation is a bug, it should be getting the new velocity for retriggered notes.

i didn't find any way to control adsfr parameters with whatsoever input. i'd like to be able to control the parameters i.e. with cc's, velocity, or with lfo's or even other env modules. again, i might be doing something wrong.

It's not implemented at the moment but it is on the todo list.

when using an adsfr on the multifilter's cutoff, i wasn't able to modulate the ammount with whatsoever cc or velocity. is there a possibility to connect a source to an ammount, as soon as one connected a source previously? i.e. as soon as i connected an adsfr to the cutoff, i assumed to see this connection-ammount to be availlable in the multifilter's destination menue...

As a general feature whereby any input's depth control could be modulated by another input or controller, it's not very easy to do. But there are a couple of options for achieving the same ends in specific situaions like for the ADSFR and Multifilter cutoff...

1. A level modulation input on the ADSFR, i have been intending to do this anyway.

2. A cutoff modulation modulation input. So it'd just be an extra input to the MultiFilter, when nothing is pluged in to the "cutoff modulation modulation" it would behave as it does now, when something is pluged in, it would act as an extra multiplier for the cutoff modulation signal.

i see no way to connect a multifilter right behind the voice mixer. i wanted the multifilter to be monophonic, using a static highpass, as the pulsewave produces a _lot_ of bass. in order to remove the hefty bass content i only need a monophonic filter. is there a way?

Not ATM. An EQ effect module is on the todo list though.

you have a "constant" source availlable in the modulators. could you maybe add another one called "alternate"? it would just send 2 values, -1 and 1. every note on would switch beween those 2 values. when i.e. connected to the voice panning, one could spread the voices evenly in the panorama, as each keystroke would pan each voice oppositely. that would be great.

I have an idea for a module with a list of user defined values and then have either random select on note on, step thru them, or time sequenced.

So certainly it would do what you describe.

while we're at the sources, could you maybe add a "random" source to the modulators? it would send out one random value per keystroke. i know i could abuse an lfo for that, but for a simple radomizing per note on this probably would be more efficient than using an lfo.

Actually i was going to include that with the recent midi sources feature, but it turned out to be a fair bit more work that I first thought as the signal input objects dont currently get retrigger events. So it is on the todo list.

the "pitch" (keyfollow) in the cc module has a e3(?) as fixed base note, which makes it impossible to use it on the spread of the supersaw. it's basically needed to detune more on lower note, but less on higher notes (sounds more natural). without beling able to specify the base note, when using the "pitch" as a source for the supersaw spread, the spread is hax (specified ammount value) at the lowest as well as at the highest note, getting narrow in the middle, where c3 does null it. so i vote for being able to freely adjust the base note when choosing the "pitch" parameter as a source. there's hundreds of other cases where this would be useful.

Its midi note 64.

For the suppersaw you have seperate picth and frequency detune anyway, so you can use the herz detune to get less "beating" on higher notes. I generally find detune the lowest key by herz, then tweak the highest one by pitch, works best.

But if you cant get it to do what you want I'll look into it again. And I wasnt entirely happy with the two seperate detune controls anyway. Always felt a bit clunky to me.

It might actually be better to enable a custom keymapping of the detune.

Actually I need to have a bit more of a think about note pitch, I see your point about it zeroing in the middle. Hmm.

the keytracking in the multifilter is unipolar upwards only. i suggest you to make it bipolar, so that one can have negative values, the higher a note is played. i know i could use "pitch" as a source, but why waste a module, if the parameter essentially only has to be changed to bipolar? also here, an adjustable base note would be really helpful.

i would love to see a bypass switch on modules. sometimes i'd want to i.e. bypass the filter, or the extra osc that discretely modulated the cutoff at audiorate, i wouldn't want to be forced to always have to set the ammunt to 0.

why different modules for the osc's? i'd rather put a dropdown menue entry for the different osc modes. that way i don't have to delete > instanciate > recable every time, just because i choose to use another osc type.

that's it for now.
i have to say it again, in case all my "moaning" leaves a different picture - this is highly addictive. i'm really not impressed easily nowadays, due to the nature of my job, but thisone is a bright light, really. love it!

I'll have to come back to the last few paragraphs as it's getting late and I have work in the morning. But I do apreciate all your input / sugestions.

Many thanks,

[gpunk]

Couple of small additions that i am badly missing right now (Might be me not seeing it though)
1 BPM synced LFO
2 Nameable modules (So i haven't got ADSFR 1/2/3/4 and so on in the lists but something i will recognise immediately)

I have a bunch of wavetables i have drawn that i will send you once i have converted em over

[brok landers]

at first: as i see it you reuse voices. i.e. retriggering the same note uses the same voice. while this is great for cpu saving, it has one flaw that i've yet to be able to overcome (maybe i'm doing something wrong). when using velocity on the cutoff of a multifilter, the velocity stays the same as it was on the first note on, when retriggering the same note while the same voice still is in amp release stage. so, is there a way to turn off the voice reusing (of course at the expense of more cpu usage)? or how do i go about that issue?

Is it that you specifically want to not reuse voices, or just that you want the fresh velocity to appear on retriggered notes? The former situation was designed to be like that because I dont see how you can reliably know which noteoff goes which which noteon if notes overlap. The later situation is a bug, it should be getting the new velocity for retriggered notes.

wel,, both. of course i'd like to have the velocity to be retriggered at its correct value always. but the reusing of voices often doesn't sound as it should. imagine a chord that you trigger staccato, it has a long filter env release and a long amp env release. now, while the filter env makes the cutoff come down slowly, you'd want that movement to be continued, regardless if you retrigger the _same_ chord. but when reusing voices, as soon as i hit the same chord again while the previous one is still in release stage, the filter env movement (as well as the amp env movement) is interrupted. so, essentially, i'd make it switchable, preferably in the master module. make a switch called "vr" (voice reuse). if it's ticked, the voice management behaves like it is currently, if it's off, the voice management will make sure that always a free, unused voice is triggered, regardless of which note is pressed. depending on the polyphony one sets up of course.

i didn't find any way to control adsfr parameters with whatsoever input. i'd like to be able to control the parameters i.e. with cc's, velocity, or with lfo's or even other env modules. again, i might be doing something wrong.

It's not implemented at the moment but it is on the todo list.

crucial imo. at least it's a modular synth and the user of course expects more to be possible than in any fixed architecture synth.

when using an adsfr on the multifilter's cutoff, i wasn't able to modulate the ammount with whatsoever cc or velocity. is there a possibility to connect a source to an ammount, as soon as one connected a source previously? i.e. as soon as i connected an adsfr to the cutoff, i assumed to see this connection-ammount to be availlable in the multifilter's destination menue...

As a general feature whereby any input's depth control could be modulated by another input or controller, it's not very easy to do. But there are a couple of options for achieving the same ends in specific situaions like for the ADSFR and Multifilter cutoff...

1. A level modulation input on the ADSFR, i have been intending to do this anyway.

2. A cutoff modulation modulation input. So it'd just be an extra input to the MultiFilter, when nothing is pluged in to the "cutoff modulation modulation" it would behave as it does now, when something is pluged in, it would act as an extra multiplier for the cutoff modulation signal.

on 1.:
that often would lead to not being able to make multiple use of the env. f.e. if i use the env on the cutoff and the velocity should control the ammount of the env, and at the same time i need this env on the resonance, but without the velocity (and with another ammount), this would be not possible. i would have to waste another env on this task, which would not only very fast clutter the patch, but also lead to unnecessary cpu consumption.

on 2.:
this is simply not the same. if i modulate the env ammount, the end value of the parameter controlled stays the same and the start value is modulated. if i modulate the cutoff itself with the velocity, the opposite is true - the end value is modulated, but the starting value of the controlled parameter stays fix. big difference.

both is needed of course, it depends on what the sounddesigner wants to achieve.

i see no way to connect a multifilter right behind the voice mixer. i wanted the multifilter to be monophonic, using a static highpass, as the pulsewave produces a _lot_ of bass. in order to remove the hefty bass content i only need a monophonic filter. is there a way?

Not ATM. An EQ effect module is on the todo list though.

an eq module is cool too. i suggest to implement a 4 band parametric eq with the following specs:
- border bands switchable from bell to shelf to cut, all with adjustable q (very important)
- the two mid bands as bells with adjustable q
- +-24db
- all bands range from 10hz up to 22khz
- frequency resolution of the knobs at least in steps of 10hz

this can be an ordinary biquad of the easiest kind, any clever music dsp rbj will do, if it's clever programmed.
see, your synth has a _very_ heavy low end. which is really good for a lot of tasks. but often one needs to make sure, especially due to the modular aproach where hardcore frequencies could come up, that he can painstakingly limit the output frequency range at the very end, before the fx (or everywhere in between the voice mixer and the output module).

please see next post for the rest, i exeeded the maximum of nested quotes here...

[brok landers]

part two:

you have a "constant" source availlable in the modulators. could you maybe add another one called "alternate"? it would just send 2 values, -1 and 1. every note on would switch beween those 2 values. when i.e. connected to the voice panning, one could spread the voices evenly in the panorama, as each keystroke would pan each voice oppositely. that would be great.

I have an idea for a module with a list of user defined values and then have either random select on note on, step thru them, or time sequenced.

So certainly it would do what you describe.

great. more than i was suggesting.

while we're at the sources, could you maybe add a "random" source to the modulators? it would send out one random value per keystroke. i know i could abuse an lfo for that, but for a simple radomizing per note on this probably would be more efficient than using an lfo.

Actually i was going to include that with the recent midi sources feature, but it turned out to be a fair bit more work that I first thought as the signal input objects dont currently get retrigger events. So it is on the todo list.

great, thanks!

the "pitch" (keyfollow) in the cc module has a e3(?) as fixed base note, which makes it impossible to use it on the spread of the supersaw. it's basically needed to detune more on lower note, but less on higher notes (sounds more natural). without beling able to specify the base note, when using the "pitch" as a source for the supersaw spread, the spread is hax (specified ammount value) at the lowest as well as at the highest note, getting narrow in the middle, where c3 does null it. so i vote for being able to freely adjust the base note when choosing the "pitch" parameter as a source. there's hundreds of other cases where this would be useful.

Its midi note 64.

For the suppersaw you have seperate picth and frequency detune anyway, so you can use the herz detune to get less "beating" on higher notes. I generally find detune the lowest key by herz, then tweak the highest one by pitch, works best.

But if you cant get it to do what you want I'll look into it again. And I wasnt entirely happy with the two seperate detune controls anyway. Always felt a bit clunky to me.

It might actually be better to enable a custom keymapping of the detune.

Actually I need to have a bit more of a think about note pitch, I see your point about it zeroing in the middle. Hmm.

well, whatever way you'll come up with will be fine, though i think the easiest and predictable way would probably still be the possibility to adjust the base key.

the keytracking in the multifilter is unipolar upwards only. i suggest you to make it bipolar, so that one can have negative values, the higher a note is played. i know i could use "pitch" as a source, but why waste a module, if the parameter essentially only has to be changed to bipolar? also here, an adjustable base note would be really helpful.

i would love to see a bypass switch on modules. sometimes i'd want to i.e. bypass the filter, or the extra osc that discretely modulated the cutoff at audiorate, i wouldn't want to be forced to always have to set the ammunt to 0.

why different modules for the osc's? i'd rather put a dropdown menue entry for the different osc modes. that way i don't have to delete > instanciate > recable every time, just because i choose to use another osc type.

that's it for now.
i have to say it again, in case all my "moaning" leaves a different picture - this is highly addictive. i'm really not impressed easily nowadays, due to the nature of my job, but thisone is a bright light, really. love it!

I'll have to come back to the last few paragraphs as it's getting late and I have work in the morning. But I do apreciate all your input / sugestions.

Many thanks,

don't worry, take your time.

[brok landers]

ok, here's some other things:

- when trying to modulate the resonance with the modwheel i realized that i can't have negative values. so if the resonance is fully up, i can't lower it by moving the modwheel upwards. in essence, all unipolar sources should have a bipolar ammount.

- in the osc's a very nice thing is to be able to restart the phase with each new note on. however, even better would be a knob instead of a switch, where i can set the phase freely in 360 degrees (basically like in the lfo's the phase knob). does wonders on basses, as well as it does when frequency modulating in audible range.

- when using keyfollow (pitch) as source from the cc module and route it to the lfo speed, and max the ammount, the keyfollow is not in tune (not a complete octave). even if the lfo is slow enough to do so.

- i miss a triangle and a sine on the osc's.

- is there any manual addon that describes how to edit the wavetable osc? i.e. is it drawable, or does it take the table only from a section of samples?

- is the wavetable osc able to actually play wav's? that would allow to buld the best drumsample ever... Smile

more to come...
----
reg, the brok

[brok landers]

some more:

- the noise module restarts the phase, as well as it outputs the exact same "figure" when reduced to crackle. so with every note pressed at the same time (like a quanized chord), the crackle "figure" is the exact same for each note.
that's also why there's no stereo sound whatsoever, when using it in unison mode - it just doubles. while this behave is just fine for some tasks, for others it's not good. could you maybe add a switch to freerun (or randomize the startpoint)?

[sonigen]

- when trying to modulate the resonance with the modwheel i realized that i can't have negative values. so if the resonance is fully up, i can't lower it by moving the modwheel upwards. in essence, all unipolar sources should have a bipolar ammount.

That's fixed.

- in the osc's a very nice thing is to be able to restart the phase with each new note on. however, even better would be a knob instead of a switch, where i can set the phase freely in 360 degrees (basically like in the lfo's the phase knob). does wonders on basses, as well as it does when frequency modulating in audible range.

Added to the ideas list.

- when using keyfollow (pitch) as source from the cc module and route it to the lfo speed, and max the ammount, the keyfollow is not in tune (not a complete octave). even if the lfo is slow enough to do so.

It should track perfectly, i'll look into it.

- i miss a triangle and a sine on the osc's.

You can get triangle from the Syncrosc by using square shape and integration set to full. You can get sine from the default wavetable at position 0.75.

That said enough people say it that I think I need to add those anyway.

- is there any manual addon that describes how to edit the wavetable osc? i.e. is it drawable, or does it take the table only from a section of samples?

It's under "Wavemaker" in the help file, maybe I should add it to the wavetable section, or add a link to it.

Right now it only works with samples, but I want to add more features to it.

To be honest while it makes some cool wavtables it didnt turn out to be as good as i thought it would due to the fact that it's actually quite hard to get a smooth progression of harmonics out of a sample. I have a few ideas to investigate, but i have too little time atm.

- is the wavetable osc able to actually play wav's? that would allow to buld the best drumsample ever...

No. It actualy extracts harmonic snapshots and generates a series of single cycle waveforms. Starts with a sample and ends up as a Waldorf type wavetable oscillator.

But a sample playback oscillator will arrive at some point.

- the noise module restarts the phase, as well as it outputs the exact same "figure" when reduced to crackle. so with every note pressed at the same time (like a quanized chord), the crackle "figure" is the exact same for each note.

It's supposed to randomize on triggering, will fix.

Thanks!

[brok landers]

and some more:

- you could maybe think about giving the supersaw section a possibility to adjust the count of saws needed... not always one needs the full count, sometimes less is more, especially when using it with unison.

- while we're at the supersaw... you could drastically enhance the "super" function in the way that you could add this finction to any osc in general, also to the wavetable. so one can use that "multiplying" on whatever waveform. very powerful.

[sonigen]

and some more:

- you could maybe think about giving the supersaw section a possibility to adjust the count of saws needed... not always one needs the full count, sometimes less is more, especially when using it with unison.

- while we're at the supersaw... you could drastically enhance the "super" function in the way that you could add this finction to any osc in general, also to the wavetable. so one can use that "multiplying" on whatever waveform. very powerful.

The count for the supper saw is easy to do, not so easy to supersize the other oscillators. Well I could do it the easy way and just duplicate the code, but it will cost 5x as much CPU for a super 5x wavetable. The supersaw is a special case whereby an extra saw only costs about 15% more in cpu.

It might be easier to make a super wavtable without the wavtable sweeping, so it's just a single cycle waveform. I need to dig into the code and see how it would work out.

Thanks,

[rob_lee]

Hello Chris,

Lovely synth and i agree with everything Brok has mentioned here, good advice IMO, i read you About Us page Made me laugh a bit "Ok it's only me" haha.
Anyways im away on hols shortly but when you have something nearly ready for release im sure i could stick some presets in there for you along with a few others i know here.. Free of course
I like the synth very much, sounds great, can't wait to hear the finished version.

Cheers

Rob

[mkastrup]

First things that came to mind for me was as Brok said, Phase and EQ. This is super important, cant do basses and drums stuff without it

Ahh oops and a smooth random LFO, nice to have if you plan to make smooth random crossovers between oscillators.

Ex. patch here that needs a smooth random LFO

http://www.xsynth.com/download/Smooth_Ran_LFO.smp

BTW. PAN on voicemixer got wrong direction, pan left goes to my right speaker

/Michael

[Ingonator]

Agreed, this seems to be a great synth.

There is also a thread about it at the Instruments forum:
http://www.kvraudio.com/forum/viewtopic.php?t=323564

I have already done some comments there (i just found this forum here...) so i would not repeat everything again.
The features which i currently miss are more FXs (e.g. Chorus, Reverb, Phaser), a step sequencer and an Arpeggiator. I am also missing an oscillator for Triangle and Sine waveforms.

I see there are already tons of suggestions here so i guess that my suggestions are already mentioned maybe.

Like Rob Lee already offered i'll try too add some presets too if i find time.

BTW will the full version be free and if not what will be the price?


Ingo

[davidka]

I liked the first impression. This synth sounds comparable to the G2 demo, at least to me. I'd suggest to have the choice of cabling with mouse.
Apart from this, I don't I can serve you with sane, constructive suggestions right now...

[brok landers]

First things that came to mind for me was as Brok said, Phase and EQ. This is super important, cant do basses and drums stuff without it

Ahh oops and a smooth random LFO, nice to have if you plan to make smooth random crossovers between oscillators.

Ex. patch here that needs a smooth random LFO

http://www.xsynth.com/download/Smooth_Ran_LFO.smp

BTW. PAN on voicemixer got wrong direction, pan left goes to my right speaker

/Michael

ahh, yes, michael is right, the marked... i totally forgot to mention thatone (did i?)... to be precise a "random'n glide" waveform is really needed, it's a real basic, and extremly often necessary. it's basically the same as a random waveform, exept that it glides from one value to another, rather than "switches" to it.