🚀 go-pugleaf

From article <2li1ar$7vs@linus.mitre.org>, by abramsh@maestro.mitre.org (Howard Abrams):
> In article <xktru*Ey1@dynam.adsp.sub.org>, jmpor@dynam.adsp.sub.org (Jean-Marc Porchet) writes:
> |> In article <2laeps$dik@news.cs.tu-berlin.de>, Dirk Kuhlmann writes:
> |> 
> |> 
> |> > 
> |> > The problem:
> |> > 
> |> > Audio data of 8kHz, 8 bit ulaw shall be converted to 44.1kHz 16 bit linear
> |> > representattion. There is no DSP support. The conversion is part of a
> |> > multi-channel audio mixing driver and therefore has to be as fast as
> |> > possible. A FFT is not possible due to timing restrictions.
> |> 
> |> You can forget to do a "good quality" sample rate converter without a
> |> DSP.(I asked our company's signal processing guru and his answer was that
> |> there is no clean way that is also a fast way (ie. you need a lot of cycles
> |> to do a clean job).
> |> 
> 
>  I missed the original post, but why do you even need a transform?
> If you want to convert 8Khz audio data to 44.1Khz, but don't care if
> it sounds like 8Khz, why not just repeat samples? that will convert
> the data to 44.1Khz, but it will have the same effect as if you
> played it back a 8Khz with the original sample.

'cause without filtering to below 4KHz you will get _vast_ quantities of
hash from the sampling.  As I said before, you have to filter the 8KHz
stuff (analogue ar digital) and then resample if you want it to look
like 44KHz samples.
Allan Duncan		ACSnet	 a.duncan@trl.oz
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