Re: [music-dsp] Time-variant 2nd-order sinusoidal resonator

2019-02-20 Thread Ian Esten
The problem you are experiencing is caused by the fact that after changing the filter coefficients, the state of the filter produces something different to the current output. There are several ways to solve the problem: - The time varying bilinear transform:

Re: [music-dsp] FFT for realtime synthesis?

2018-10-23 Thread Ian Esten
The Kurzweil K150 is the first product I can think of that did it. To create custom sounds for it required the use of software that modeled the sound using partial amplitudes over time. It's a very powerful technique for synthesising certain types of sound, such as a piano, where frequencies of

Re: [music-dsp] tracking drum partials

2017-07-30 Thread Ian Esten
That would also be my first choice. It might also be worth looking at modern algorithms in that family of methods. A lot of effort has gone into designing exponentially damped sine methods for voice compression and transmission. They will be more robust to noise than Prony's method. Some methods

Re: [music-dsp] Transient shaping - differential envelope?

2016-07-06 Thread Ian Esten
Look up spectral flux. On Wed, Jul 6, 2016 at 7:24 AM, Danijel Domazet < danijel.doma...@littleendian.com> wrote: > Hi music-dsp, > How does one implement an envelope adjustment algorithm that is triggered > only on transients, rather than on a loudness threashold which is used in > conventional

Re: [music-dsp] Announcement: libsoundio 1.0.0 released

2015-09-06 Thread Ian Esten
This discussion is a refreshing change from some recent topics. Constructive, respectful, not insulting. This is how it should be. On Sun, Sep 6, 2015 at 2:41 AM, Ross Bencina wrote: > Hello Andrew, > > Thanks for your helpful feedback. Just to be clear: I maintain

Re: [music-dsp] Announcement: libsoundio 1.0.0 released

2015-09-04 Thread Ian Esten
I was going to ask the same question, until I looked at the webpage. The features are listed out nicely. On Fri, Sep 4, 2015 at 9:58 AM, Brad Fuller wrote: > On 09/04/2015 09:42 AM, Andrew Kelley wrote: > > libsoundio is a C library providing cross-platform audio input and

Re: [music-dsp] Announcement: libsoundio 1.0.0 released

2015-09-04 Thread Ian Esten
Thanks for sharing. Looks nice! A question: I see that the write callback supplies a minimum and maximum number of frames that the callback is allowed to produce. I would prefer a callback that instructed me to produce a given number of samples. It is simpler and more consistent with existing

Re: [music-dsp] Uses of Fourier Synthesis?

2015-04-05 Thread Ian Esten
On Sun, Apr 5, 2015 at 3:32 PM, robert bristow-johnson r...@audioimagination.com wrote: On 4/5/15 5:21 PM, Theo Verelst wrote: In the context of synthesis, or intelligent multi sampling with complicated signal issues, you could try to make the FFT analysis and filtering a targeted part of the

Re: [music-dsp] Linearity of compression algorithms on more than one sound component

2015-02-13 Thread Ian Esten
Lossy encoding wouldn't necessarily be non-linear in all cases. Of course it is non-linear. Lossy encoding does not satisfy the conditions of linearity: f(a + b) = f(a) + f(b) f(a.b) = a.f(b) -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive,

Re: [music-dsp] Linearity of compression algorithms on more than one sound component

2015-02-12 Thread Ian Esten
It's lossy. Definitely not linear. On Thu, Feb 12, 2015 at 4:33 PM, robert bristow-johnson r...@audioimagination.com wrote: On 2/12/15 3:02 PM, Theo Verelst wrote: Hi all, Just a thought I share, because of associations I won't bother you with, suppose you take some form of audio

Re: [music-dsp] Thoughts on DSP books and neural networks

2015-02-05 Thread Ian Esten
Octave or Matlab. Or even Mathematica. It would be very interesting to see the transfer function of your filter on the same graph as the 'ideal' analog filter. Ian On Thursday, February 5, 2015, Peter S peter.schoffhau...@gmail.com wrote: What do you guys use to turn your impulse responses

Re: [music-dsp] note onset detection

2013-08-08 Thread Ian Esten
resolution and increase the density to get better time resolution when you find a note on. That would be pretty low cost. On Thu, Aug 8, 2013 at 11:23 AM, robert bristow-johnson r...@audioimagination.com wrote: On 8/8/13 11:05 AM, Ian Esten wrote: On Mon, Aug 5, 2013 at 1:01 PM, robert bristow

Re: [music-dsp] who else needs a fractional delay.

2010-11-19 Thread Ian Esten
A Leslie emulation (or effect similar to that) might well need one, depending on how you modeled it. Same statement applies for tape delay style effects too. As you say, I bet there's plenty of others, too. Anyone else got any other effects to add to the list? Ian On Fri, Nov 19, 2010 at 1:07