Arturia V Collection 9 - Official Thread

[rezoneight]

Check out TAL J-8 for another reference point. I think it sounds quite a bit better than the Arturia version but they’re just analog synth emulations. With FX off you can say they’re all pretty plain Jane

Its interesting that he added some stuff like velocity and being able to sync either OSC (vs the original) but didn't provide the polarity switch on ENV-1. Wonder why that choice. On the other hand he's got the dual/split mode and Arturia decided to get rid of that for some reason and it seems that just having two instances running isn't quite the same thing. Will check it out.

[Stefken]

@Bmanic, Is oversampling your hobby ?
Btw. You can create a feedback loop in the ms -20 (out into the signal processor and back in) which gives it more oumph.

[waltercruz]

Check out TAL J-8 for another reference point. I think it sounds quite a bit better than the Arturia version but they’re just analog synth emulations. With FX off you can say they’re all pretty plain Jane

Its interesting that he added some stuff like velocity and being able to sync either OSC (vs the original) but didn't provide the polarity switch on ENV-1.

The ENV MOD can be positive or negative, there is the polarity switch.

[rezoneight]

Check out TAL J-8 for another reference point. I think it sounds quite a bit better than the Arturia version but they’re just analog synth emulations. With FX off you can say they’re all pretty plain Jane

Its interesting that he added some stuff like velocity and being able to sync either OSC (vs the original) but didn't provide the polarity switch on ENV-1.

The ENV MOD can be positive or negative, there is the polarity switch.

I do not see that in the manual but haven't looked at the actual plug-in yet. No polarity switch mentioned there.

[Funkybot's Evil Twin]

I do not see that in the manual but haven't looked at the actual plug-in yet. No polarity switch mentioned there.

It's a bi-polar fader. So if you want negative modulation you just pull the Env Depth fader below the center point and you're into negative territory. Note: this is the same on the Roland Cloud versions and System-8.

[rezoneight]

I do not see that in the manual but haven't looked at the actual plug-in yet. No polarity switch mentioned there.

It's a bi-polar fader. So if you want negative modulation you just pull the Env Depth fader below the center point and you're into negative territory. Note: this is the same on the Roland Cloud versions and System-8.

Ohhh...you're talking about the fader under VCO Modulator. Got it. Wow didn't even realize that when looking at the manual. Loaded up the plugin and was looking around thinking "WTF is this guy talking about..." thank you!

Looks like thats another thing he added then since the LFO fader also goes negative.

[kvotchin]

As far as I know, Piano V has always been modeled and not sampled, and the same goes for all the organs and electric pianos. I believe the only V Collection instruments that use samples are:

• Emulations of sample-based synths, such as Emulator II, SQ80, and CMI (Fairlight).
• Mellotron, where samples are used in place of tape loops (but processed to sound like tape).
• Vocoder, for the mode that uses triggerable samples as the modulator instead of live audio input.
• The new Augmented Strings & Voices plugins.

Thanks.

Actually, came across some interesting info since asking that, re: Pianoteq and Arturia Piano V. Turns out, Piano V 2/3 in fact use older(?) Pianoteq code! I had no idea.

[bmanic]

@Bmanic, Is oversampling your hobby ?

My hobby and profession involves critical listening. Thus the understanding of when and when not to use oversampling is quite often a focus. Let's just put it this way: If more people would understand this, I'd have a lot less clients that are suffering from harsh and "honky" mixes (no matter how much they try to scoop things in the 1-5kHz region).

Btw. You can create a feedback loop in the ms -20 (out into the signal processor and back in) which gives it more oumph.

That is a different kind of oumph.. and, yes I know how to use the MS-20 thank you.

[bmanic]

Actually, came across some interesting info since asking that, re: Pianoteq and Arturia Piano V. Turns out, Piano V 2/3 in fact use older(?) Pianoteq code! I had no idea.

No wonder they sound a bit weird. I really like them for all the creative stuff.. but as an actual piano? Nope. Not even close (especially the low octaves.. there are some really problematic weird resonances there).

[EnGee]

Can the oversampling fix the aliasing after it is done in the plugin? I thought the oversampling is just to avoid the aliasing from happening inside the plugin, but once it happens, the oversampling after that is useless. Am I wrong?

[Stefken]

I d think once the ‘mirror’ frequencies are in the signal, any oversampling is not going to change that, but i ll let the oversampling experts answer that...

[EnGee]

Yes! Once, Urs (U-he) explained it to me in an old thread. In short, if the coder hasn't applied oversampling, then you can do nothing to minimize the aliasing. At least this is what I understood!

[bmanic]

Can the oversampling fix the aliasing after it is done in the plugin? I thought the oversampling is just to avoid the aliasing from happening inside the plugin, but once it happens, the oversampling after that is useless. Am I wrong?

Once you have aliasing, you have aliasing. Period. There is no way to remove it.

Thus it is important to understand where and when it happens so that you can mitigate it.

Again, as an example:

1) Digital oscillators, 12bit, low sample rate -> aliasing happening.. very audible especially at high frequencies.

2) --> goes into analogue Low Pass filter -> diminishes all aliasing from the cutoff frequency forward.

3) if the filter is a digital model of an analogue filter, with non-linearities, it will cause additional harmonic distortion which will alias if not prevented to do so by the developer. This harmonic distortion will further interact with the oscillators mentioned at point 1).

Thus you will need to oversample the filter emulation (or use some other clever means as a developer) to minimize further aliasing from the harmonic distortion caused by the filter. Especially if you want the emulation to be authentic as no aliasing of said filter will happen in the analogue domain.

[bmanic]

Yes! Once, Urs (U-he) explained it to me in an old thread. In short, if the coder hasn't applied oversampling, then you can do nothing to minimize the aliasing. At least this is what I understood!

That is incorrect. If the plugin allows you to work with it at high sample rates, the aliasing will never happen (provided that YOU do the oversampling AND filtering.. or use a 3rd party or DAW solution that does it for you).

However, if there is a chain of non-linearities where some of the chain's components alias within the actual plugin, hitting the other non-linearities.. and it will stubbornly be at some predefined samplerate internally, no matter what external sample rate you are running, THEN there is absolutely nothing you can do about it.

.. but I've never actually heard of such poor design by any developer ever and never encountered that kind of scenario.

It is worth noting that quite a few plugins do not properly work at anything higher than 96kHz. This is poor design by the developer and is unfortunately quite common.

[bmanic]

Also one more point: Aliasing is not "bad" per say. If that is the sound you are going for.

Example:

A really lo-fi, hugely aliasing and bit reduced sample. Would fit any chip tune perfectly. However, now consider the case where you want to further process this sample with an overdrive stomp box. In the analogue domain this will nicely crunch up the sample and change the tonality predictably. Try to do the same thing in the digital domain with a bad overdrive stomp box emulation that causes further aliasing and the tonality is completely different from the analogue version. Thus you have a choice now, a fork in the road. You can try to massively oversample the stomp box emulation to get the tonality you would have gotten from the analogue stomp box. Or perhaps you really like the additional aliasing that is caused by the poor stomp box emulation?

If it's the latter, then you are all good and can ignore the "problem". If it's the former, you need the oversampling.. in which case more is better (or "closer" to the analogue stomp).

Oversampling isn't a free lunch though. The very process of up and down sampling + filtering will cause small changes to the sound, but usually these changes are much smaller than the actual aliasing itself.