[linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions

[Denis Sbragion]

Hello Michael,

At 10.13 28/07/2003 +0200, you wrote:
...
>A 22050 Hz sine could be really accurate (one sample up, one sample down
>every 1/22050th second), and so is 11025. But intermediary frequencies
>introduces temporal aliasing, some metallic feeling due to temporal
>quantization. This is inherent to the very low sampling rate (96 kHz is just
...


sorry, but this is incorrect. The sampled signal has nothing to do with the 
information it carries. The sampled signal just contains the information 
needed to get back the original signal but it isn't the original signal 
(despite at lower frequency it pretty much resemble the original signal), 
you have to pass it through a reconstruction filter to get the original 
(bandlimited) signal back. You can't simply look at the sampled signal to 
see what the original signal was, much the same way you can't just look at 
a compressed file to see what the original file was. This is one of the 
most common misconception circulating about PCM, but it is indeed a 
misconception.
         If you hear a difference between DSD and PCM the problem should be 
somewhere else, may be in the ADC/DAC used, may be in the DSD itself, but 
sure not in what you describe above.

Bye,
--
	Denis Sbragion
	InfoTecna
	Tel: +39 0362 805396, Fax: +39 0362 805404
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