[linux-audio-user] Sample Rate Conversions (was: Converting sample rate: failed...)

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Was: Converting sample rate: failed...

Erik,

Regarding your recent post on this subject:

> Funnily enough the issue you see as a "serious misconception" I see
> as a "significant advantage of libsamplerate over your converter". [0]

> [0] You claim that the highly localized behavior of the truncated
>     windowed sinc is a bad thing. I claim that this localization is
>     a good thing for converting a general digital audio signal because
>     these signals are already highly localized (ie snare drum hit at 
>     10.03 seconds from the start). From your description, your 
>     converter spreads these highly localized events over the whole 
>     of the output signal which I think is a bad thing.

-----------------------------------------

This is absolute nonsense.  Your misconception is even more deeply
rooted than I had thought.  You are obviously missing some theoretical
background I had assumed you had.  If you had had this background, you
probably wouldn't have written the rest of your post, in addition to
not having posted this nonsense.

You mentioned an interest in certain characteristics of the resampling
method I'm using.  Either do the analysis or do the literature research.
I've already told you everything you need to know: FFT with and without
overlap; overlap accomplished with raised cosine windows.  Only someone
who is utterly clueless would insist on obtaining my code for measurements.
(And only someone who is very immature would characterize this standard
advice as "hot air" before they even saw it.)

Good luck,
Dave.



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