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Applied mathematics / Sampling / Low-pass filter / Finite impulse response / Fourier transform / Electronic filter / Nyquist–Shannon sampling theorem / Filter bank / Digital signal processing / Signal processing / Electronic engineering
Date: 2014-06-09 10:18:15
Applied mathematics
Sampling
Low-pass filter
Finite impulse response
Fourier transform
Electronic filter
Nyquist–Shannon sampling theorem
Filter bank
Digital signal processing
Signal processing
Electronic engineering

2006 Paper 9 Question 10 Digital Signal Processing Consider a software routine that converts the sampling rate of digital audio data from 8 kHz to 48 kHz, without changing the represented sound. It reads an input sequen

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