DAC interpolation 
DAC interpolation 
Mar 2 2013, 03:25
Post
#1


Group: Members Posts: 2 Joined: 2March 13 Member No.: 106971 
The NyquistShannon theorem (the sampling theorem) states that if a function x(t) contains no frequencies greater than B hertz, it is completely characterized by measuring its amplitudes at a series of points spaced 1/(2B) seconds apart. It further states that to reconstruct the original signal one must use the WhittakerShannon interpolation theorem, computing the sum of the infinite series of normalized sinc functions for each sample. (Well,
CODE $x(t)=\sum_{\infty}^{\infty}x[n]\cdot{\rm sinc}(\frac{tnT}{T})$ where n is the sample number, t is the sample time, T is 1/(sampling rate), and sinc(x) is the normalized sinc function.)I've never (in my somewhat limited experience) seen an audio DAC that does that, yet they seem to have pretty reasonable output. How accurate are the standard approximations, really? What is the interpolation error? How does one characterize it? 


Mar 4 2013, 04:10
Post
#2


Group: Members Posts: 2 Joined: 2March 13 Member No.: 106971 
Thanks, I think I understand a bit better now. Especially how the use of a lowpass filter approximates the use of the sinc function in the sampling theorem.



LoFi Version  Time is now: 30th March 2015  11:17 