Re: [music-dsp] Dynamic smoothing algorithm

2016-12-07 Thread robert bristow-johnson
Original Message Subject: Re: [music-dsp] Dynamic smoothing algorithm From: "Lubomir I. Ivanov" Date: Wed, December 7, 2016 5:13 pm To: "A discussion list for music-related DSP" -

Re: [music-dsp] Dynamic smoothing algorithm

2016-12-07 Thread Lubomir I. Ivanov
hello, On 6 December 2016 at 08:26, Andrew Simper wrote: > Hi Guys, > > Another year has almost passed so I thought it was time to release > another technical paper! > > It's a dynamic smoothing algorithm that can do things like this: > > http://cytomic.com/files/dsp/dynamic-smoothing.png > > I c

Re: [music-dsp] Allpass filter

2016-12-07 Thread Stefan Stenzel
> On 7 Dec 2016, at 13:10 , Uli Brueggemann wrote: > > Hi, > > I'm searching a solution for an allpass filter calculation with following > conditions: > > There is a given pulse response p with a transfer function H. It is possible > to derive a linear phase pulse response lp from the magnit

Re: [music-dsp] Allpass filter

2016-12-07 Thread Evan Balster
> > *The magnitude of ap = 1, so ap applies only phase shifts. Its group delay > is inverse to the group delay of p.* > I think the idea here is to derive a filter with the same magnitude characteristics as p, but zero group delay --- so as to transform arbitrary filters into zero-phase / zero-del

Re: [music-dsp] Allpass filter

2016-12-07 Thread Ethan Duni
I'm not sure I quite follow what the goal is here? If you already have lp and p, then there aren't any additional calculations needed to obtain ap - it's an IIR filter with numerator coefficients given by lp, and denominator coefficients given by p. The pulse response is obtained by running the fil

Re: [music-dsp] Allpass filter

2016-12-07 Thread Stefan Sullivan
A linear phase all-pass filter is a delay. Stefan On Dec 7, 2016 4:30 AM, "STEFFAN DIEDRICHSEN" wrote: > > On 07.12.2016|KW49, at 13:10, Uli Brueggemann > wrote: > > Is there a solution to elegantly calculate the pulse response ap ? The > calculation of p^-1 may be difficult or numerically uns

Re: [music-dsp] Allpass filter

2016-12-07 Thread STEFFAN DIEDRICHSEN
> On 07.12.2016|KW49, at 13:10, Uli Brueggemann > wrote: > > Is there a solution to elegantly calculate the pulse response ap ? The > calculation of p^-1 may be difficult or numerically unstable. A spectral inversion can be a challenging at times. However, lp/p should be fine to calculate,

Re: [music-dsp] Allpass filter

2016-12-07 Thread Remy Muller
sounds like a good candidate for Wiener deconvolution: https://en.wikipedia.org/wiki/Wiener_deconvolution On 07/12/16 13:10, Uli Brueggemann wrote: Hi, I'm searching a solution for an allpass filter calculation with following conditions: There is a given pulse response p with a transfer fun

[music-dsp] Allpass filter

2016-12-07 Thread Uli Brueggemann
Hi, I'm searching a solution for an allpass filter calculation with following conditions: There is a given pulse response p with a transfer function H. It is possible to derive a linear phase pulse response lp from the magnitude of H. Now there is an equation p * ap = lp (* = convolution, ap =