I have seen that clip before. I have made some comparisons myself and the JP8000 sounds a bit better IMHO than the virus for that "late 90s big detuned trance lead" sort of stuff. I am not sure exactly why that is, might be the high pass filter at the output, or the way JP uses prime numbers, while virus has symmetric detune, or the noise osc or the aliasing...

Yeah, the diva recreation seems spot on. The hypersaw is a different beast from the supersaw though. It's not really a supersaw copy.

The point of my original post was to address the first question that is usually asked when creating a supersaw. How many saw waves are wanted? First thing is of course to figure out at what point adding more saws does nothing new... That's kind of what the post was about.

Fair enough. It won't get much further then the answer you provided yourself:

"trust your ears then"

Do you prefer the sound of the unison or hypersaw, btw?

Don't think there's any right and wrong when approaching this. It's right when it feels/sounds right, so tweak away and don't be such a tweak head :twisted:

I think you can choose 48k sample rate on the Virus ti, no? this should, presumably, push the aliasing a bit further to the right of the spectrum, making it less noticeable. Interestingly enough, the Virus has much larger definition then the good old JP does. A slight boost on the highs can help to.

I was going to take it easy, but no.

For the sake of truth: that's a pure 30.5Hz wave, it's got more then one complete cycle, in fact there's 30 and a half. The only thing you can complain about is "discontinuity" - so go ahead and search for that.

Further more: we can ear sine waves and sine waves don't actually produce any sort of clip in their beginning, not unless reproduction of the audio starts or ends on a non zero crossing point. You may have that "feeling" with instruments whose oscillators are incapable of fixing phase start position. You can set phase initial position on the Virus, btw.

The only thing wrong with my sample is that you can't loop it and still get a perfect 30.5Hz sine wave, you'd have to go back, choose a zero cross point at the end of a cycle, and then if you were to loop that, you'd have a perfect 30.5Hz sine wave playing for as long as it makes you happy. Sampling frequency (the time you set to analyse a signal) has nothing to do with the frequency of the signal itself, so as a matter of fact you are INDEED confusing a lot of things. Mentioning spectral leakage is almost laughable at this point, since you show absolutely no accuracy in your remarks and you're failing to provide a solid basis for any of your claims. In fact, it all reads like pretentious rubbish talk to me and any informed reader. So there's a little honesty for you to digest slowly with a pinch of salt m8.

And how do you fit more than one complete cycle in the one second window? You can't. It's not a sine wave. Easier way to think about this is trying to fit a 0.5hz sine into the window. Now it's obvious that you have a rectified sine wave on your hands, which has entirely different spectrum than a sine wave.


This is incorrect. We cannot hear a real sine wave (just try it by setting the osc to 0 phase- you will hear the click and can even EQ it). A real sine wave would have to be infinite. When you play a shorter than that sine wave it's combined with the gate(silence) at the start. This kind of function will have to have a different spectrum than a pure sine wave and it will contain (almost) all frequencies - some of which are cut off by the converter's reconstruction filter altering the sine wave. A perfectly round finite sine wave can't even be reproduced...

If you loop a 30.5hz sine that is cut off at 1 second mark, you would hear exactly what I have stated - the spectrum would contain more than one frequency and the frequency of the wave would be some multiple of the 1hz. 30.5hz would be obviously impossible inside a 1 second window. Although our hearing works very differently from a spectrum analyzer so using a much smaller window works much better - try 0.01sec. The point that you can slice a wave and loop it past the window has nothing to do with any of this.

I can't? I just did. I've made the seemingly trivial task of placing a 30,5Hz inside a second time frame for you and - once again - that gives you 30 complete cycles and a half. Do yourself a favour and open it in an audio editor software and check for yourself.

P.S.

With the Virus setting "phase init" to 0 means there's absolutely no phase initial position at all. That's in the manual btw.

No you didin't. Watch this (most important part starts around 3:07):


This demonstrates that if sine is tuned to frequency that can't exist in the window - the impossible sine frequency doesn't magically emerge from the spectrum. Rather the signal consists of many different allowed frequencies. That is also called spectral leakage. 30.5hz frequency can not exist in a one second window, rather such wave would contain different allowed frequencies of 30hz, 31hz, and maybe other frequencies...

Do you understand that half a cycle sine wave IS NOT a sine wave anymore? This is where your confusion seems to stem from. Sinewave has to start and end at the same phase, if it doesn't - it's not a sine wave. And that is what the FFT shows (it's a collection of different sine waves).

From the comments bellow:

"NTS Press1 year agoin reply to Alex Wong Chin Yung

Yes, any situation that causes the array to contain anything other than an integer number of sinusoidal periods will result in leakage."

that is the case with this sample I posted. it translates to discontinuity. discontinuity is what's causing the sample leakage you seem so obsessed about. plus, as I've stated previously, frequency domain for the FFT analysis is simply the time you choose to pick a sample, in other words. thus, one way to compensate for that, besides choosing integral numbers of the frequency of the signal - which is the same as looping one cycle btw, but I'm not even going to explain why to you - is fading in and out, 'cause if the waveform meets 0 crossing at both ends, then there's continuity and FFT is only going to pick the exact frequencies contained on the signal. How does this work for you? Happy?

But the software on that link is actually quite useful, so thanks for pointing it out.

https://meocloud.pt/link/c794215c-b3...018.34.02.png/

that's how I produced the sample, and that's the oscilloscope showing no signs of disturbance, plus an FFT analysis of the frequency spectrum showing a single harmonic in there. FFT resolution isn't big enough to show a perfect line on low frequencies for reasons - again - I'm not going to explain to you.

This turned out to be even a bigger mess then I thought it was. So as of now, I rest my case. Cheers. :cool:

I don't see how the picture you produced is relevant to anything you or I have said. We are talking about 30.5hz sine wave existing in a ONE SECOND WINDOW here. I have never said that 30.5hz sine wave is impossible to be produced in a completely arbitrary window, which is what you used. (Or span at the sample length of, is that around 32k samples? - which failed at showing a point as well). In fact, 2 second window has a frequency resolution of 0.5hz - no problem producing 30.5hz there.

I think in the last post you made you yourself admitted that the length of the sample determinates the frequency resolution. Isn't that what this was all about?

Yes, it's determined by that. But FFT is just a way of analysing a signal. The frequency resolution is how long (in time) that sample is - aka as block size. In other words, we agree on that.

What I don't agree with is that there's no single complete cycle within that array. Because there is. And if you were to loop just one of them, the result would be exactly like that in the picture that I posted. So it's possible to produce a 30.5Hz sine wave within the constraints of that time span. If it was to end half a cycle earlier, the frequency of the signal would be the same, there would be no audio click, just a little silence before it starts over. And absolutely no sample leakage.

The problem is that the samples collected for the FFT analysis are not synced to the frequency of the signal, it's dependant on the refresh rate you set for it and the number of samples it collects (block size again), so there's no perfect alignment between the two things - which need to be for producing accurate results instead of displaying waveforms cut at random points that will indeed be interpreted (calculated) as a different waveform altogether and hence show some other harmonics which should not be there. So if you were to fade in and out that array, discontinuity would not be a problem, no other waveform would be calculated instead of that present in the signal, but you'd need a fairly high refresh rate to compensate for the measurement of amplitude, presumably - not even pretending to be an expert here.

But why does this matter so much to you? Even with all this in mind, let's assume we're on the same page as of now - that I finally made some sense of your words - how do you explain why note duration would be such an important thing for supersaws?

P.S. on span I've just selected high resolution, didn't manually select block size. ;)