🚀 go-pugleaf

In <2li1ar$7vs@linus.mitre.org> abramsh@maestro.mitre.org (Howard Abrams) writes:
>In article <xktru*Ey1@dynam.adsp.sub.org>, jmpor@dynam.adsp.sub.org (Jean-Marc Porchet) writes:
>|> In article <2laeps$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.


Very bad idea, you will get a lot of aliasing.

Because using this methode you will play your 8KHz samples @ 8KHz, you
would need a 4KHz corner-freq. anti-aliasing filter !
That would be very bad all the rest of your signal (the part that need
44.1KHz)

So get your 8KHz signal to 44.1 inserting 0 between the samples and then
apply a digital low-pass filter with 4 KHz corner freq. Finally you can
mix-it with the 44.1KHz signal.

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