[Asterisk-Dev] Re: Handling AST_FORMAT_SLINEAR at 48000Hz

[voipwala hindustani]

Hi Kevin, Luigi and others,
Thanks for your discussion about this problem.

Kevin/Luigi - Can you suggest me some open source implementation of "first
order low-pass filter".
I tried using libresample-0.1.3 and Secret Rabit code. But they dont give me
a clean API and are cryptic. Need to go through all the code to understand
their parameters they need and how to set them. I am doing this at the
moment, but if you have come across some other library which can give me a
cleaner interface, please suggest.

Thanks and Regards,
Voipwala

On 11/17/05, Tony Mountifield <tony at softins.clara.co.uk> wrote:
>
> In article <20051117024659.E88354 at xorpc.icir.org>,
> Luigi Rizzo <rizzo at icir.org> wrote:
> > On Thu, Nov 17, 2005 at 10:28:47AM +0000, Tony Mountifield wrote:
> > >
> > > The low-pass filter sounds complicated. How would that compare with,
> say,
> > > taking the mean average of every six 48kHz samples to produce one 8kHz
> > > sample?
> >
> > it's just not the same thing and will give rise to all sort of
> > frequency aliasing effects - those that the lowpass filtering is
> supposed
> > to remove.
> > quick example: consider a 9khz signal - that is clearly not audible
> > on a 4KHz channel (8000 hz sampling rate). Yet if you sample it at
> > 48khz and take the average of every 6 samples, these averages will not
> > be constant (because the 9khz signal does not have a period of 6
> samples)
> > and so you'll get an output signal that will appear as 1KHz.
> > This is the aliasing effect.
>
> That makes sense - thanks for the explanation!
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
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