Reply by Robert Lietzmann February 28, 20072007-02-28
>Hi! > >I hope I'm not disturbing. >Maybe this is also interesting for you:
If you want to know how to use CZT for auditory analysis of speech/music signals, please have a look to following article: An auditory spectral analysis model using the chirp z-transform Kates, J.; Acoustics, Speech, and Signal Processing [see also IEEE Transactions on Signal Processing], IEEE Transactions on Volume 31, Issue 1, Feb 1983 Page(s):148 - 156 Summary: In this paper we discuss a new signal-processing approach to implementing an auditory processing model. The model is based on auditory physiology and psychophysics; it uses constant-bandwidth analysis below 500 Hz and constant-Q analysis above 500 Hz..... Have a nice day, Robert
> >> >>"Andor" <an2or@mailcircuit.com> wrote in message >>news:ce45f9ed.0403131537.34ec7122@posting.google.com... >>> Matt Timmermans wrote: >>> Although what you write is of course correct, this method doesn't >>> strike me as very optimal in any sense. You have to take care of
block
>>> effects when processing streams, it has high code complexity (FFT vs. >>> FIR or polynomial interpolation), and fractional sample rate changes >>> come at extremely high processing cost (as opposed to polyphase >>> filtering approach). >> >>Yes, there are lots of situations in which that method of resampling is >>inapplicable or impractical. >> >>> Isn't it the other way round: One uses FFTs to calculate chirp-z >>> transforms? >> >>Both ways: You chirp-z to do a weird-length FFT. The chirp-z is based >on >>convolution, which you implement with a power-of-2-length FFT/IFFT. >> > > > >I could be wrong, but isn't 'weird-length FFT' some contradiction? Isn't >this a regular DFT, and a power-of-2-DFT is an FFT and the the >chirp-z-transformation can be used to calculate some, let's call it, >N-to-M DFT, with some N+M+1 < 2^k-FFT? > > > >>> I think the chirp-z transform has no real-world application in >>> resampling. Parameters like quality and processing time are much >>> better controlable with polyphase filtering or polynomial >>> interpolation, at much lower code complexity. >> >>I would tend to think the same thing, except that I remember reading >more >>than one thread about FFT-based resampling in this very NG. >> >> >> >> > >best regards, >Thomas >
Reply by quaste February 8, 20072007-02-08
Hi!

I hope I'm not disturbing.
..

> >"Andor" <an2or@mailcircuit.com> wrote in message >news:ce45f9ed.0403131537.34ec7122@posting.google.com... >> Matt Timmermans wrote: >> Although what you write is of course correct, this method doesn't >> strike me as very optimal in any sense. You have to take care of block >> effects when processing streams, it has high code complexity (FFT vs. >> FIR or polynomial interpolation), and fractional sample rate changes >> come at extremely high processing cost (as opposed to polyphase >> filtering approach). > >Yes, there are lots of situations in which that method of resampling is >inapplicable or impractical. > >> Isn't it the other way round: One uses FFTs to calculate chirp-z >> transforms? > >Both ways: You chirp-z to do a weird-length FFT. The chirp-z is based
on
>convolution, which you implement with a power-of-2-length FFT/IFFT. >
I could be wrong, but isn't 'weird-length FFT' some contradiction? Isn't this a regular DFT, and a power-of-2-DFT is an FFT and the the chirp-z-transformation can be used to calculate some, let's call it, N-to-M DFT, with some N+M+1 < 2^k-FFT?
>> I think the chirp-z transform has no real-world application in >> resampling. Parameters like quality and processing time are much >> better controlable with polyphase filtering or polynomial >> interpolation, at much lower code complexity. > >I would tend to think the same thing, except that I remember reading
more
>than one thread about FFT-based resampling in this very NG. > > > >
best regards, Thomas
Reply by Matt Timmermans March 14, 20042004-03-14
"Andor" <an2or@mailcircuit.com> wrote in message
news:ce45f9ed.0403131537.34ec7122@posting.google.com...
> Matt Timmermans wrote: > Although what you write is of course correct, this method doesn't > strike me as very optimal in any sense. You have to take care of block > effects when processing streams, it has high code complexity (FFT vs. > FIR or polynomial interpolation), and fractional sample rate changes > come at extremely high processing cost (as opposed to polyphase > filtering approach).
Yes, there are lots of situations in which that method of resampling is inapplicable or impractical.
> Isn't it the other way round: One uses FFTs to calculate chirp-z > transforms?
Both ways: You chirp-z to do a weird-length FFT. The chirp-z is based on convolution, which you implement with a power-of-2-length FFT/IFFT.
> I think the chirp-z transform has no real-world application in > resampling. Parameters like quality and processing time are much > better controlable with polyphase filtering or polynomial > interpolation, at much lower code complexity.
I would tend to think the same thing, except that I remember reading more than one thread about FFT-based resampling in this very NG.
Reply by Andor March 13, 20042004-03-13
Matt Timmermans wrote:
...
> "One Usenet Poster" wrote: > > DSP Gurus: > > > > I'm familiar with the classic interpolate-filter-decimate approach to > > multirate DSP. I've heard that the chirp-z transform can be used to > > resample a signal also. Can anyone provide additional information or > > references (other than Google search results)? > > Well, one well-known technique for interpolation is to do an FFT, pad the > signal with zeros in the frequency domain, and then use the IFFT to get the > interpolated signal.
Although what you write is of course correct, this method doesn't strike me as very optimal in any sense. You have to take care of block effects when processing streams, it has high code complexity (FFT vs. FIR or polynomial interpolation), and fractional sample rate changes come at extremely high processing cost (as opposed to polyphase filtering approach).
> Using the chirp-z transform to perform FFTs and IFFTs > with arbitrary lengths would make it convenient to do this for odd sampling > rate ratios, as in converting between 44100 and 48000 Hz, or stretching > audio by small amounts to keep a video sync.
Isn't it the other way round: One uses FFTs to calculate chirp-z transforms? And a fractional rate change like 44100 -> 48000 would require an upsampling by factor 160 first, followed by downsampling by a factor of 147. This means you have to take an IFFT of the signal at a sampling rate of 7 MHz. Hardly convenient for real-time processing.
> That's not a terriffic advantage, but off the top of my head, I can't think > of any other way to use chirp-z for resampling.
I think the chirp-z transform has no real-world application in resampling. Parameters like quality and processing time are much better controlable with polyphase filtering or polynomial interpolation, at much lower code complexity. Regards, Andor
Reply by Matt Timmermans March 12, 20042004-03-12
"One Usenet Poster" <me@my.computer.org> wrote in message
news:1052b8jauoi1417@corp.supernews.com...
> DSP Gurus: > > I'm familiar with the classic interpolate-filter-decimate approach to > multirate DSP. I've heard that the chirp-z transform can be used to > resample a signal also. Can anyone provide additional information or > references (other than Google search results)?
Well, one well-known technique for interpolation is to do an FFT, pad the signal with zeros in the frequency domain, and then use the IFFT to get the interpolated signal. Using the chirp-z transform to perform FFTs and IFFTs with arbitrary lengths would make it convenient to do this for odd sampling rate ratios, as in converting between 44100 and 48000 Hz, or stretching audio by small amounts to keep a video sync. That's not a terriffic advantage, but off the top of my head, I can't think of any other way to use chirp-z for resampling.
Reply by Bhaskar Thiagarajan March 12, 20042004-03-12
"One Usenet Poster" <me@my.computer.org> wrote in message
news:1052b8jauoi1417@corp.supernews.com...
> DSP Gurus: > > I'm familiar with the classic interpolate-filter-decimate approach to > multirate DSP. I've heard that the chirp-z transform can be used to > resample a signal also. Can anyone provide additional information or > references (other than Google search results)? > > Thanks, > OUP
Read the latest (or perhaps a month old) Embedded Programming Magazine. You can find an online version at www.embedded.com. It had a nice article on this topic for the uninitiated. BTW, I'm not sure if I'd say that the Chirp-Z is useful for resampling. An alternative to the FFT as well as for some specific applications like time-frequency analysis is what I'd describe it's use in general terms. Cheers Bhaskar
Reply by One Usenet Poster March 11, 20042004-03-11
DSP Gurus:

I'm familiar with the classic interpolate-filter-decimate approach to 
multirate DSP.  I've heard that the chirp-z transform can be used to 
resample a signal also.  Can anyone provide additional information or 
references (other than Google search results)?

Thanks,
OUP