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Sampling and Interpolation Yao Wang Polytechnic University, Brooklyn, NY11201 http://eeweb.poly.edu/~yao
Outline • Basics of sampling and quantization – A/D and D/A converters
• Sampling – Nyquist sampling theorem – Aliasing due to undersampling: • temporal and frequency domain interpretation • Sampling sinusoid signals
• Reconstruction from samples – Reconstruction using sample-and-hold and linear interpolation – Frequency domain interpretation (sinc pulse as interpolation kernel)
• Sampling rate conversion – Down sampling – Up sampling – Demonstration
How to determine T and Q? • T (or fs) depends on the signal frequency range – A fast varying signal should be sampled more frequently! – Theoretically governed by the Nyquist sampling theorem • fs > 2 fm (fm is the maximum signal frequency) • For speech: fs >= 8 KHz; For music: fs >= 44 KHz;
• Q depends on the dynamic range of the signal amplitude and perceptual sensitivity – Q and the signal range D determine bits/sample R • 2R=D/Q • For speech: R = 8 bits; For music: R =16 bits;
• One can trade off T (or fs) and Q (or R) – lower R -> higher fs; higher R -> lower fs
• We consider sampling in this lecture, quantization in the next lecture
Nyquist Sampling Theorem • Theorem: – If xc(t) is bandlimited, with maximum frequency fm (or ωm =2π fm ) – and if fs =1/T > 2 fm or ωs =2π /T >2 ωm – Then xc(t) can be reconstructed perfectly from x[n]= xc(nT) by using an ideal low-pass filter, with cut-off frequency at fs/2 – fs0 = 2 fm is called the Nyquist Sampling Rate
• Physical interpretation: – Must have at least two samples within each cycle!