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Consider a D/A converter for audio signals consisiting of a zero-order-hold interpolator followed by a continuous-time lowpass filter with positive passband between 0 and 20KHz and stopband starting at fa= 40KHz.

Assume we want to convert a digital signal originally sampled at 16KHz. What is the minimum oversampling factor that we need to use?

for getting 16 khz i used 2 as my oversampling factor which seems not right. can you elaborate on top of it ?? i used 2 because (40khz/20khz)

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    $\begingroup$ exact same problem as your previous question: this is homework, dumped on us, without showing any own attempt. We don't do that here. $\endgroup$ Sep 13 '20 at 9:37
  • $\begingroup$ for getting 16 khz i used 2 as my oversampling factor which seems not right. can you elaborate on top of it ?? i used 2 because (40khz/20khz) $\endgroup$
    – dsp
    Sep 13 '20 at 9:54
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    $\begingroup$ ... edit your question to include that, instead of posting it as a comment. $\endgroup$ Sep 13 '20 at 9:54
  • $\begingroup$ done sir. Hope to get reply soon:) $\endgroup$
    – dsp
    Sep 13 '20 at 10:05
  • $\begingroup$ Does this answer your question? D/A converter with $\endgroup$ Sep 13 '20 at 21:16

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