Re: [LAU] octaver (plugin) for bass

From: Gianfranco Ceccolini <gianfranco@email-addr-hidden>
Date: Tue Jun 10 2014 - 18:56:29 EEST

Sorry for the late posting, but I’d also recommend the mod-pitchshifter

https://github.com/portalmod/mod-pitchshifter

We have developed 4 plugins: Capo ( 1 to 7 semitones up), the SuperCapo (1 to 24 semitones up), Drop (1 to 12 semitones down) and SuperWhammy (continuous travel from -12 to 24 semitones)

They are a bit CPU hungry but sound quality is quite good.

Hope I’ve helped

Gianfranco
The MOD Team

Em 10/06/2014, à(s) 12:27, rosea grammostola <rosea.grammostola@email-addr-hidden> escreveu:

> I'm not sure if real hardware stompboxes of this type are better then this plugin, but this might be the kind of sound I might like to buy a hardware stompbox for. Maybe I can test one somewhere and compare it with Gxdetune. Thanks for the comments Fons, tips for improvement are always welcome I think.
>
>
> On Sun, Jun 8, 2014 at 1:34 PM, Fons Adriaensen <fons@email-addr-hidden> wrote:
> On Sun, Jun 08, 2014 at 12:36:17PM +0200, hermann meyer wrote:
>
> > Without downsampling it use (well, 4xtimes more then now) 8% dsp
> > load. Most costs in the original source comes from that used values
> > are not pre-calculated.
> > But indeed, the reason for downsampling is that the limited
> > frequency range makes it sound good,
>
> because that removes most of the broadband junk that would be generated
> otherwise...
>
> > and for guitar/bass 3kHz are
> > far more then enough when you would add a octave up/down to the
> > original sound.
>
> True for bass and guitar.
>
> Still this algorithm is far from what it could be. I don't blame
> for you that, it's Bernsee who is missing the consequences of his
> own analysis (which is valid as far as it goes).
>
> Take alook at his table labeled 'pass #5'. The input signal is
> halfway between two bins. Assume we want one octave up. The expected
> output signal corresponds exactly to bin 225. For that signal, the
> output of the analysis FFT would be (similar to 'pass #1):
>
> bin amplitude
> ------------------
> 223 0.000
> 224 0.500
> 225 1.000
> 226 0.500
> 227 0.000
>
> And that is of course also what the correct input to the synthesis
> IFFT should be. Which is quite different from what the algorithm
> produces (by scaling each bin individually):
>
> bin amplitude
> -------------------
> 222 0.170
> 223 0.000
> 224 0.849
> 225 0.000
> 226 0.849
> 227 0.000
> 228 0.170
>
> The result of this after the IFFT is the correct frequency, but
> with two periods of the window applied (it will be zero at the
> center).
>
> The frequency values that are calculated provide exactly the
> information required to avoid this and to do the correct
> calculation. But it's just thrown away.
>
> Ciao,
>
> --
> FA
>
> A world of exhaustive, reliable metadata would be an utopia.
> It's also a pipe-dream, founded on self-delusion, nerd hubris
> and hysterically inflated market opportunities. (Cory Doctorow)
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user@email-addr-hidden
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user@email-addr-hidden
> http://lists.linuxaudio.org/listinfo/linux-audio-user

_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@email-addr-hidden
http://lists.linuxaudio.org/listinfo/linux-audio-user
Received on Tue Jun 10 20:15:02 2014

This archive was generated by hypermail 2.1.8 : Tue Jun 10 2014 - 20:15:02 EEST