[linux-audio-user] Converting sample rate: failed...

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On Tue, 14 Sep 2004 12:33:14 -0700
davidrclark@xxxxxxxxxxxxx wrote:

> Erik,
> 
> I do not low-pass filter in my resampler, yet it works fine. 

You're doing the resampling in thre frequency domain, right? You
FFT the data, fiddle with the frequency domain data and then 
inverse FFT it to get back to the time domain.

Even though you aren't explicitly applying a filter you are
very likely to be doing frequency domain lowpass filtering.

<snip>

> I use 
> a raised cosine window in the time domain and no window (other than the
> rectungular truncation "window" for the case of downsampling) in the 
> frequency domain. 

OK, so as a (very) simplified example, say you are downsampling 
by a factor of 2. You FFT N data points to get N data points in 
the frequency domain, you throw away N/2 samples in the middle
and then inverse FFT to get back to the time domain.

Notice that step where you throw away the middle N/2 samples?
That is a low pass filter applied in the frequency domain.

Erik
-- 
+-----------------------------------------------------------+
  Erik de Castro Lopo  nospam@xxxxxxxxxxxxx (Yes it's valid)
+-----------------------------------------------------------+
The difference between genius and stupidity is that
  genius has its limits.

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