Re: [linux-audio-dev] Resampling

2001-08-17 Thread Jussi Laako
Alexander Ehlert wrote: > > other than exponents of 2. The problem is, that the so called > split radix routines can be really slow. So I'm not actually sure Split radix isn't slow in itself, in fact it's usually faster that radix-2/4/8 transforms on architectures with large multilevel caches.

Re: [linux-audio-dev] Resampling

2001-08-14 Thread Juhana Sadeharju
>From: Jussi Laako <[EMAIL PROTECTED]> > >I've got lot of stability problems with high order IIR lowpass filters. >Those tend accumulate significant DC offsets over time. OK for short samples >but not for realtime streaming, IMHO. I were reading about the stuff from Jon Dattorro's paper in Journ

Re: [linux-audio-dev] Resampling

2001-08-13 Thread Jussi Laako
Juhana Sadeharju wrote: > > I'm about to examine these float/double issues for IIR filters in any > day now. I just wrote an IIR core as well. I've got lot of stability problems with high order IIR lowpass filters. Those tend accumulate significant DC offsets over time. OK for short samples but

Re: [linux-audio-dev] Resampling

2001-08-09 Thread Juhana Sadeharju
>From: Alexander Ehlert <[EMAIL PROTECTED]> > >Why 4096 FFT array?? I just use any blocksize. FFTW supports blocksizes >other than exponents of 2. The problem is, that the so called You would make a good patent lawyer because you would now be able to grand a patent for any other FFT array length

Re: [linux-audio-dev] Resampling

2001-08-09 Thread Alexander Ehlert
Hi, > For sample rate converter (not for sampling synth interpolator), > FIR filter is a must because they can be made linear-phase easily. > There are near linear-phase IIR filters which could be used, but > I have not yet read about them that much that I could write code > for you. You can do

Re: [linux-audio-dev] Resampling

2001-08-08 Thread Juhana Sadeharju
>From: Alexander Ehlert <[EMAIL PROTECTED]> > >> AFAIK almost none uses the sinc interpolation since it is too cpu hungry. >> Most softynths work without any filtering otherwise they would not achieve >> 50-100 voices on a PII/III box. > >Yeah, you would a sinc interpolation with folding anyway a

Re: [linux-audio-dev] Resampling

2001-08-08 Thread Alexander Ehlert
Hi Benno, > Alex, I'm not sure if I understood the functioning of your resampler correctly: > can you specify only integer factors or fractional values too ? by increasing the blocksize of the fft, you can virtually do any resample frequency. OK an example, we use a 2048 samples fft. This gives

Re: [linux-audio-dev] Resampling

2001-08-02 Thread Juhana Sadeharju
>From: Richard Dobson <[EMAIL PROTECTED]> > >I suspect there are other aspects too where the implementation couyld be >improved. More orthodox pvocs, sucha s the F.R Moore one, and CARL (on >which my 'PVStream' real-time pvoc is based), does offer exact >reconstruction. Because of the interpolati

Re: [linux-audio-dev] Resampling

2001-08-02 Thread Richard Dobson
The Csound pvoc has many differences from others, not least in that it offers time-scaling by means of interpolating between the analysis frames of the pre-created analysis file. It is not a simple streaming pvoc the way others are. I suspect there are other aspects too where the implementation co

Re: [linux-audio-dev] Resampling

2001-08-02 Thread Alexander Ehlert
Hi, > OK. I have been quite skeptical to any FFT analysis/synthesis because > the phase vocoder in Csound sounds very muddy even no change is done. > But if you're handling only the complex values got from FFT (not amplitude- > phase versions), then everything should be better. For my ears it so

Re: [linux-audio-dev] Resampling

2001-08-02 Thread Nick Bailey
Juhana Sadeharju wrote: > OK. I have been quite skeptical to any FFT analysis/synthesis because > the phase vocoder in Csound sounds very muddy even no change is done. Something's wrong. Overlap? Window shape? > But if you're handling only the complex values got from FFT (not amplitude- > ph

Re: [linux-audio-dev] Resampling

2001-08-01 Thread VEdran Vucic
You can try MIDIMAN cards too. please visit www.midiman.com vedran Vucic On Wednesday 01 August 2001 09:00, you wrote: > >Is there any card, that works fine with full duplex 44kHz and small > >fragment sizes? Not the Hammerfall, I want something cheaper. > > I use a Trident 4D-NX based card

Re: [linux-audio-dev] Resampling

2001-08-01 Thread Juhana Sadeharju
>From: Alexander Ehlert <[EMAIL PROTECTED]> > >I just do an n-times oversampling DFT. You can configure blocksize >and oversampling(overlap) factor. Then for the resampling process >I just add/remove bands in the frequency domain and then do resynthesis. >Thats it. In the oversampling and resynth

Re: [linux-audio-dev] Resampling

2001-08-01 Thread Paul Davis
>Is there any card, that works fine with full duplex 44kHz and small >fragment sizes? Not the Hammerfall, I want something cheaper. I use a Trident 4D-NX based card from Hoontech that cost about $60, and it works 100% with 64 frames per interupt. --p

[linux-audio-dev] Resampling

2001-08-01 Thread Alexander Ehlert
Hi Juhana, sorry with all that gui discussion I totally forgot about this email. > What exactly is the algorithm? Does it do a good job? How have you > tested it? I just do an n-times oversampling DFT. You can configure blocksize and oversampling(overlap) factor. Then for the resampling process