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

[davidrclark at earthlink.net]

Erik,

I do not low-pass filter in my resampler, yet it works fine.  The reason
is that I assume that the input is band-limited, and this is usually true
for my own work.  Not only is it band-limited, but usually also tapered
in the frequency domain, i.e. already effectively low-pass filtered.  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.  Effectively I am assuming oversampling in the time 
domain.  I also have filtering capability, but separately from resampling,
for cases in which I do have higher frequencies.  (However, this is also
very crude from a user point of view...)

On the presence or absence of higher frequencies:  If there is further
processing down the road and the higher frequencies are missing, then the
results may be inaccurate (for example absent cross-products in the audible
region which are not the result of aliasing).  Throwing out high frequencies,
or merely altering them somehow at every stage is not necessarily advisable,
so I caution people who extol the virtues of low-pass filtering.  Now in 
a library resampler, such as yours, I'm sure it's a good idea to enable it...
I wouldn't volunteer to answer your email otherwise!

Best regards,
Dave.