Unfiltered Audio Lion

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dionenoid wrote: Wed Sep 18, 2019 2:56 pm
Phil B wrote: Wed Sep 18, 2019 1:09 pm Not disrespecting your opinion at all, just curious ... what soft synths do you place at the other end of the spectrum (ie great and full of life)?
Spire, Serum, Avenger, Diva, Falcon, Massive X and don't forget Sylenth1 and Synthmaster.

For capabilities and flexibility of routing, fx and osc's i would say Melda SoundFactory and Falcon but also Synthmaster and Avenger are miles ahead of Lion. I have yet to demo Phaseplant but on paper that one looks to be a better choice also.

But hey, it's just an opinion. Everyone has to decide for themselves ofcourse, so just demo and judge for yourself. If you're into this kind of sound and/or programming than you'll probably love it.
:tu: Thx

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dionenoid wrote: Wed Sep 18, 2019 2:56 pm
Phil B wrote: Wed Sep 18, 2019 1:09 pm Not disrespecting your opinion at all, just curious ... what soft synths do you place at the other end of the spectrum (ie great and full of life)?
Spire, Serum, Avenger, Diva, Falcon, Massive X and don't forget Sylenth1 and Synthmaster.

For capabilities and flexibility of routing, fx and osc's i would say Melda SoundFactory and Falcon but also Synthmaster and Avenger are miles ahead of Lion. I have yet to demo Phaseplant but on paper that one looks to be a better choice also.

But hey, it's just an opinion. Everyone has to decide for themselves ofcourse, so just demo and judge for yourself. If you're into this kind of sound and/or programming than you'll probably love it.
It's a matter of taste for sure - I have Falcon, Massive X, and Avenger (& I like them all) but they don't do the same things for me that Lion does. It's way easier to dial in extreme sound-design stuff in Lion than in these other synths, which is one of the first things I look for; but I agree that's not necessarily what everyone wants in an instrument.

One other random note about Lion: I just discovered that if you set the Byome comb filter to quantize mode, turn the frequency down to the minimum, and route keyboard pitch tracking to frequency with maximum modulation, the comb filter will exactly follow the input pitch (but only monophonically since the Byome effects aren't polyphonic). This gives some really cool physical-modeling style results if you run the noise sources (eg Dust) through it. It's probably too soon for a wish list but I would love to see the comb filter as one of the (polyphonic) filter types! (I know that there's also the resonator bank but it would be neat to have a polyphonic comb too.)

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Definitely a matter of taste, and Lion is certainly to my personal taste - like a hi-res, advanced Intellijel Shapeshifter with loads of modulation. Was just curious how others viewed “the other side of the coin”

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dionenoid wrote: Wed Sep 18, 2019 2:56 pm
Phil B wrote: Wed Sep 18, 2019 1:09 pm Not disrespecting your opinion at all, just curious ... what soft synths do you place at the other end of the spectrum (ie great and full of life)?
Spire, Serum, Avenger, Diva, Falcon, Massive X and don't forget Sylenth1 and Synthmaster.

For capabilities and flexibility of routing, fx and osc's i would say Melda SoundFactory and Falcon but also Synthmaster and Avenger are miles ahead of Lion. I have yet to demo Phaseplant but on paper that one looks to be a better choice also.

But hey, it's just an opinion. Everyone has to decide for themselves ofcourse, so just demo and judge for yourself. If you're into this kind of sound and/or programming than you'll probably love it.
Thanks for the detailed response...hope I didn't sound too snarky before!

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dionenoid wrote: Wed Sep 18, 2019 2:56 pm
Phil B wrote: Wed Sep 18, 2019 1:09 pm Not disrespecting your opinion at all, just curious ... what soft synths do you place at the other end of the spectrum (ie great and full of life)?
Spire, Serum, Avenger, Diva, Falcon, Massive X and don't forget Sylenth1 and Synthmaster.

For capabilities and flexibility of routing, fx and osc's i would say Melda SoundFactory and Falcon but also Synthmaster and Avenger are miles ahead of Lion. I have yet to demo Phaseplant but on paper that one looks to be a better choice also.

But hey, it's just an opinion. Everyone has to decide for themselves ofcourse, so just demo and judge for yourself. If you're into this kind of sound and/or programming than you'll probably love it.
You should test phaseplant. It is (to me at least) miles ahead from lion. The grouping, the modsources, the fx, the lay-out, the sound (!!!), the flexibilty, the routing, etc... :tu:

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WasteLand wrote: Wed Sep 18, 2019 1:59 pm i am no expert, i use the "beta" version, but it seems like that the oversampling setting (online) influences the character of the main filter, they get more drive. i analyze it, not in an expert way, there is no aliasing or not that "visible".
the sound when the filter is on gets "highs", harmonics likely. when you change oversampling. i use very simple presets, my own...
in my experience oversampling does not drastically changes the sound of the filter, i mean the difference is: at 1x: dull sound: at 2x: screams... it seems like there is more aliasing.......... but i am no expert... i program my soft-synths always from scratch, i use oversampling when necessary, but this. it makes a complete different sound.
it also gives you the feeling that you do not know what is the "correct" sound.
Increasing the oversampling rate also increases the cutoff frequency of the filter, and vice versa - decreasing the oversampling lowers the cutoff frequency. This occurs with all of the filters, except for the LP12/HP12/BP12. I don't think it's a coincidence that these are the filters that were just updated with the latest hotfix.

Hopefully UA did not plan their vacations for right after the official release of LION, as it has some serious bugs that will hopefully be addressed soon. The developers have so far seemed very responsive to the issues that have been brought to their attention in this thread. It is puzzling however that these issues were not brought to their attention during beta testing...

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I work in different project sample rates. Based on the feedback about the oversampling issues I will have to test it with different rates and see if anything else changes. Of course, it should sound the same no matter the project and oversampling rates.

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plexuss wrote: Wed Sep 18, 2019 5:14 pm I work in different project sample rates. Based on the feedback about the oversampling issues I will have to test it with different rates and see if anything else changes. Of course, it should sound the same no matter the project and oversampling rates.
On my system, changing the project samplerate had the same effect on cutoff frequency as changing the oversampling rate. I think all they need to do is apply the same fix to all of the filters as they did to LP12/HP12/BP12.

There could be other weirdness associated with samplerate too though - maybe it’s also connected to the tempo sync issues.

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regarding the envelopes.. i find them really snappy if you set the attack to exponential curve. i get great WOMP pluck like attacks that really come through on plucky type percussive sounds or metallic synthetic sounding steel drum like things. really beautiful sounding to my ears. the master ADSR has a lot of range to it. it's quite flexible so to dial it in you need to spend a moment with it to shape your sound. dial in the attack time and shape precisely and there's plenty expressiveness there for punchy sounds. at least, that's been my experience so far.

i think it's a great synth that will only get better. the oscillator models have a lot going on. i like that there are experimental type algorithms with various flavors of noise. every oscillator model might not be something you use all the time but for experimenting and finding new sounds it's a great place to look. i think it's good they included all these. i've had great results just starting with a preset and hitting the randomize button and seeing what happens.

also, the integration of the effects is fantastic. it's easy to dial in complex things with modulation and essentially make the effects part of the synthesis of the patch and do really complex sounding things. ymmv of course. but i'm sold. it's a winner for what i do.

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evo2slo wrote: Tue Sep 17, 2019 11:17 pm
Amusesmile wrote: Tue Sep 17, 2019 11:11 pm
evo2slo wrote: Tue Sep 17, 2019 9:56 pm Thanks for the hotfix. Quantization and filter working great now.

Two major issues that it would be nice to see addressed are the MIDI Input Gate not functioning with the envelope Trigger ...
Hey thanks for this video and the comments. Let's focus on this for a minute- what behavior do you want to get exactly in this video? Do you want it so that any time you hit a new note, the AD envelope fires which brings up the master output gain?
Yes. The Envelope should retrigger whenever a MIDI Input Gate is activated. The Trigger light is lighting up when a key is pressed, but the envelope is inactive as shown in the video.

edit: Just checked and it is working fine with the LFOs, retriggering them with the MIDI Input Gate whenever a key is pressed, just the ADSR and AD Envelopes seem to be not working properly as far as I can tell.
Ok what's going on is we have both a monophonic and polyphonic modulation system running in parallel so if the rules aren't fully understood it's difficult to figure out what's going on. And even then it takes me a few minutes some times to map it out in my head. There are also a few rules we didn't do a great job of communicating either in the manual or on the interface. I'll try to help with both of those and should really do a full tutorial when I have a minute. The reason we made it like this was to pack the widest range of possible behavior into the fastest/smallest footprint.

First things first the "peace sign" three lines on any output or input tell you whether that is polyphonic or not. There are rules about what modulation will result based on both the inputs and outputs:

Poly output->Mono input = modulation will be whatever the most recent note's polyphonic output will be. This will be shown as a filled area on the Mono input knob (even though it is a per-voice modulation result which would be shown as a small line were it a poly input knob).

Poly output->Poly input = independent polyphonic modulation per voice (shown by small lines)

Mono output->Mono input = modulation will follow monophonic modulation (which is a separate thing from all per-voice modulation)

Mono output->Poly input = modulation will follow monophonic modulation (resulting in a bunch of small white lines that are all in the same location)

There's an additional trigger rule which is this: any ADSR or AD envelope gets an automatic trigger any time a new note is hit for the polyphonic voice output of that modulator (assuming it's in poly mode). The reason we did this is because that's usually the behavior people want so it's quickest. If you want to prevent this (so that the poly output of each voice's AD/ADSR modulators aren't triggered at the start of the note but rather at a later time) then you can attach a cable to it. That turns this behavior off and the modulator only checks the poly input on that trigger to decide when to fire.

The "G" output on the Midi input modulator is a special thing which is basically only used to trigger the Monophonic running modulation system trigger inputs. The polyphonic version of "G" doesn't ever output anything besides 0.0. Essentially you should only use the "G" output if you want the Monophonic running modulation system to be triggered every time a new note is hit. Like resetting the sine modulator, etc. The reason it doesn't do anything Polyphonic-wise is that every single modulator already gets reset or triggered on that that polyphonic voice's new note being triggered. For instance a Sine Modulator polyphonic output is always going to reset its phase on that note when it is struck.

Sorry if this doesn't make sense- I will try to find better ways to communicate or graph it. Essentially there is an entire Monophonic modulation system running in parallel with a polyphonic one but you can get almost any behavior you want with the right combination (which is why it's so awesome/unique imo).

-----

Anyway, to get the actual behavior you want I think all you need to do is attach the AD output to the master Gain knob, DONT attach any cable to the trigger input, and then change the AD envelope to monophonic (little icon in top right) if you want a soft reset on each new note or keep it polyphonic if you want a hard reset.

Cheers!
Josh, Co-Founder of Unfiltered Audio:
http://www.unfilteredaudio.com

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That is weird, because on certain modules there is a gate trigger AND a reset trigger as well. They all "should" be triggered by the gate. At least, that is how gates normally work, right?

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Amusesmile wrote: Wed Sep 18, 2019 6:01 pm ....
Anyway, to get the actual behavior you want I think all you need to do is attach the AD output to the master Gain knob, DONT attach any cable to the trigger input, and then change the AD envelope to monophonic (little icon in top right) if you want a soft reset on each new note or keep it polyphonic if you want a hard reset.
Wow, I still haven't wrapped my head around all that but I really appreciate your taking the time to explain it. I think I confused this issue by trying address the envelope trigger and tempo sync issues in a single video. In the video I showed the envelope trigger with the Gain control just because I was using the gain control to illustrate the tempo sync issue.

My actual real world use for the envelope trigger is I was working on a mono bass sound with the ADSR envelope connected to the filter cutoff. I wanted each note I played to trigger the cutoff envelope, while holding down the root note. The behavior I was experiencing was no variation in cutoff while holding down a sustained note, even while playing other notes. I understand this behavior is sometimes desirable, but sometimes one would want each new note to trigger the envelope, even while sustaining the first note.

I made another video showing this more precisely. This time I tried switching the ADSR to monophonic but it made no difference.

Is there any way to get each new note to retrigger the filter cutoff envelope while sustaining the first note? I'm still unable to figure it out after reading your explanation LION's of modulation system.

I hope this makes sense. Thanks for your help!

https://youtu.be/LHDrTITqz0s

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Amusesmile wrote: Wed Sep 18, 2019 6:01 pm Anyway, to get the actual behavior you want I think all you need to do is attach the AD output to the master Gain knob, DONT attach any cable to the trigger input, and then change the AD envelope to monophonic (little icon in top right) if you want a soft reset on each new note or keep it polyphonic if you want a hard reset.

Cheers!
Ok, that works but it still gets out of sync if cycle is on...

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Smapti wrote: Wed Sep 18, 2019 3:16 pm
One other random note about Lion: I just discovered that if you set the Byome comb filter to quantize mode, turn the frequency down to the minimum, and route keyboard pitch tracking to frequency with maximum modulation, the comb filter will exactly follow the input pitch (but only monophonically since the Byome effects aren't polyphonic). This gives some really cool physical-modeling style results if you run the noise sources (eg Dust) through it. It's probably too soon for a wish list but I would love to see the comb filter as one of the (polyphonic) filter types! (I know that there's also the resonator bank but it would be neat to have a polyphonic comb too.)
some of the presets do this and make great use of the comb/resonator/granulator in this way. it's really quite good :)

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evo2slo wrote: Wed Sep 18, 2019 4:42 pm
WasteLand wrote: Wed Sep 18, 2019 1:59 pm i am no expert, i use the "beta" version, but it seems like that the oversampling setting (online) influences the character of the main filter, they get more drive. i analyze it, not in an expert way, there is no aliasing or not that "visible".
the sound when the filter is on gets "highs", harmonics likely. when you change oversampling. i use very simple presets, my own...
in my experience oversampling does not drastically changes the sound of the filter, i mean the difference is: at 1x: dull sound: at 2x: screams... it seems like there is more aliasing.......... but i am no expert... i program my soft-synths always from scratch, i use oversampling when necessary, but this. it makes a complete different sound.
it also gives you the feeling that you do not know what is the "correct" sound.
Increasing the oversampling rate also increases the cutoff frequency of the filter, and vice versa - decreasing the oversampling lowers the cutoff frequency. This occurs with all of the filters, except for the LP12/HP12/BP12. I don't think it's a coincidence that these are the filters that were just updated with the latest hotfix.

Hopefully UA did not plan their vacations for right after the official release of LION, as it has some serious bugs that will hopefully be addressed soon. The developers have so far seemed very responsive to the issues that have been brought to their attention in this thread. It is puzzling however that these issues were not brought to their attention during beta testing...
yes your are right those filters that you mention, do not change the character of the sound....
my sounds are almost always with a lot of harmonics (and the lion is designed for that...), so it's sometimes difficult too see where the roll of starts, the sounds are also in modulated in some way, in many cases, by different means (depends on synts, soft-synths, vintage soft-synth).

in the mean time interesting info about lion, that should be in a manual, or so...

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