Sample Rate Conversion by non integer factor

"Ronald H. Nicholson Jr." <rhn@mauve.rahul.net> wrote in message
news:cc1q5u$8k1$1@blue.rahul.net...
> In article <W6CEc.179470$207.1255211@news20.bellglobal.com>,
>
> On a fast
> PC you may not need to do this, as a well tuned math library can often
> calculate the window and Sinc tap coefficients in real time.

Just curious, how well does this work with a Kaiser window (my favorite choice
for this type of thing)?  It's got that nasty Bessel function in it!

David Reid wrote:
> 
> I have actually.  i've also heard of Open source audio library on
> sourceforge.net.  i think the implementation for the class afLibConverter
> (used for sample rate conversion) is a reimplementation of the Secret Rabbit
> Code functions in C++.

Actually, I beleive that afLibConverter uses a modified version
of Julius O. Smith's original C code and simply provides a C++
wrapper around it. 

Secret Rabbit Code is the reimplementation and last time I compared
my code (the Rabbit) with implementations based on the JOS code, my
code ran faster and had better signal to noise and a wider passband.
I've been meaning write up these results but I simply haven't had
the time.

> Anyway, I've posted the algorithm.  don't care if i can't share it. dont
> really like this company much....

Que??

Erik
-- 
+-----------------------------------------------------------+
  Erik de Castro Lopo  nospam@mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
When aiming for the common denominator, be prepared for the
occasional division by zero.

I have implemented a polynomial interpolator and it works fine for
Audio Sample rate conversion.  You can the material here
http://www.student.oulu.fi/~oniemita/DSP/INDEX.HTM

rhn@mauve.rahul.net (Ronald H. Nicholson Jr.) wrote in message news:<cc1q5u$8k1$1@blue.rahul.net>...
> In article <W6CEc.179470$207.1255211@news20.bellglobal.com>,
> David Reid <dreid_nospam@remove_no_spam_mechtronix.ca> wrote:
> >How can this be done?  My application is converting the sample rate of
> >44.1kHz wave files to sample rate X (eg change SR by factor of 0.75, or 1.3)
> >and playing back at 44.1kHz, so that the speed sounds different.  
> 
> Interpolation can give you any point between sample points, and
> thus will work for any arbitrary or even time varying sample ratios.
> 
> Sinc interpolation will produce much less distortion than linear
> or low order polynomial interpolation (see the "reconstruction"
> theorems), although linear interpolation may work if the frequency
> content of the original signal is way below the sample rate.
> 
> An interpolation method with suitable approximations for the
> windowed-Sinc function will allow you to trade off compute time against
> S/N ratio.  You can play with the type of window, width of window and
> various approximation tricks for the windowed-Sinc function.  The various
> polyphase resampling methods are just table-lookup driven optimization
> tricks which one can use with DSP's that are too slow (or as an
> optimization when needed for power reduction or something).  On a fast
> PC you may not need to do this, as a well tuned math library can often
> calculate the window and Sinc tap coefficients in real time.
> 
> 
> IMHO. YMMV.

balaji wrote:
> 
> I have implemented a polynomial interpolator and it works fine for
> Audio Sample rate conversion.  You can the material here
> http://www.student.oulu.fi/~oniemita/DSP/INDEX.HTM

Have you measured the signal to noise ratio? If so, what was
it and how did you measure it?

Erik
-- 
+-----------------------------------------------------------+
  Erik de Castro Lopo  nospam@mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
"And MS thinks Linux is vulnerable to forking? 95, 95 OEM SR2, 98, 98SE,
ME, NT, 2000, Bob, .NET, CE, Datacenter, Server, Adv. Server, and now
Web Server, sheesh." -- BTS on LinuxToday.com

Jon Harris wrote:
> "Ronald H. Nicholson Jr." wrote:
> >
> > On a fast
> > PC you may not need to do this, as a well tuned math library can often
> > calculate the window and Sinc tap coefficients in real time.
> 
> Just curious, how well does this work with a Kaiser window (my favorite choice
> for this type of thing)?  It's got that nasty Bessel function in it!

Have you tried Knab's alternative to the Kaiser window [1]? It is also
a one-parameter window family, but is defined using exp instead of the
Bessel I_0 function and is claimed to be very close to the Kaiser
window.

Regards,
Andor

[1] John J Knab, "An Alternate Kaiser Window"
IEEE Transactions on Acoustics, Speech, and Signal Processing, Vol.
ASSP-27, No.5, October 1979

"Andor" <an2or@mailcircuit.com> wrote in message
news:ce45f9ed.0407020150.26f1e6ef@posting.google.com...
> Jon Harris wrote:
> > "Ronald H. Nicholson Jr." wrote:
> > >
> > > On a fast
> > > PC you may not need to do this, as a well tuned math library can often
> > > calculate the window and Sinc tap coefficients in real time.
> >
> > Just curious, how well does this work with a Kaiser window (my favorite
choice
> > for this type of thing)?  It's got that nasty Bessel function in it!
>
> Have you tried Knab's alternative to the Kaiser window [1]? It is also
> a one-parameter window family, but is defined using exp instead of the
> Bessel I_0 function and is claimed to be very close to the Kaiser
> window.
>
> Regards,
> Andor
>
> [1] John J Knab, "An Alternate Kaiser Window"
> IEEE Transactions on Acoustics, Speech, and Signal Processing, Vol.
> ASSP-27, No.5, October 1979

That's the first I've heard of it.  Thanks for the tip!