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Chapter 5: Sampling ​

Introduction ​

Sampling is the bridge between continuous-time and discrete-time signals. Understanding sampling theory is essential for digital signal processing.

Ideal Sampling ​

Definition: Ideal Sampling

Ideal sampling multiplies a continuous signal xc(t) by an impulse train:

xs(t)=xc(t)∑n=−∞+∞δ(t−nTs)=∑n=−∞+∞xc(nTs)δ(t−nTs)

where Ts is the sampling period and fs=1/Ts is the sampling frequency.

Spectrum of Sampled Signal ​

Theorem: Frequency Domain Effect of Sampling

If xc(t)↔Xc(jΩ), then the Fourier transform of the sampled signal is:

Xs(jΩ)=1Ts∑k=−∞+∞Xc(j(Ω−kΩs))

where Ωs=2πfs=2π/Ts is the sampling angular frequency.

Proposition: Spectral Replication

Sampling creates infinite replicas of the original spectrum, centered at multiples of the sampling frequency:

  • Original spectrum centered at Ω=0
  • Replicas centered at Ω=±Ωs,±2Ωs,…

Nyquist-Shannon Sampling Theorem ​

Theorem: Nyquist-Shannon Sampling Theorem

A bandlimited signal xc(t) with maximum frequency fmax (i.e., Xc(jΩ)=0 for |Ω|>2πfmax) can be perfectly reconstructed from its samples if:

fs>2fmax

or equivalently:

Ts<12fmax

The minimum sampling rate fN=2fmax is called the Nyquist rate.

Aliasing ​

Definition: Aliasing

Aliasing occurs when the sampling rate is too low (fs<2fmax), causing spectral replicas to overlap. This results in distortion that cannot be corrected.

Example: Aliased Sinusoid

Consider sampling xc(t)=cos⁡(2πf0t) at rate fs.

If f0>fs/2, the sampled signal appears as:

x[n]=cos⁡(2πfan/fs)

where fa=|f0−kfs| is the aliased frequency (for some integer k that minimizes fa).

For example: f0=900 Hz sampled at fs=1000 Hz gives fa=100 Hz.

Normalized Frequency ​

Definition: Normalized Frequencies

For a continuous signal with frequency f (Hz) or Ω (rad/s):

Digital angular frequency:

ω=ΩTs=2πf/fs(rad/sample)

Normalized frequency:

ν=f/fs(cycles/sample)

The unambiguous range is:

  • ω∈[−π,π] or [0,2π]
  • ν∈[−0.5,0.5] or [0,1]

Ideal Reconstruction ​

Definition: Ideal Interpolation

Perfect reconstruction uses an ideal lowpass filter with cutoff fs/2:

xr(t)=∑n=−∞+∞x[n]sinc(t−nTsTs)

where sinc(u)=sin⁡(πu)πu.

Theorem: Interpolation Formula

If the Nyquist criterion is satisfied, the original signal can be recovered:

xc(t)=∑n=−∞+∞xc(nTs)sinc(t−nTsTs)

Practical Considerations ​

Anti-Aliasing Filter ​

Definition: Anti-Aliasing Filter

An anti-aliasing filter is a lowpass filter applied before sampling to remove frequencies above fs/2:

|Haa(jΩ)|≈{1|Ω|<π/Ts0|Ω|>π/Ts

Practical Reconstruction ​

Proposition: Zero-Order Hold (ZOH)

Practical DACs use a zero-order hold:

xzoh(t)=x[n]for nTs≤t<(n+1)Ts

The ZOH frequency response is:

Hzoh(jΩ)=Tssinc(ΩTs2π)e−jΩTs/2

Oversampling and Decimation ​

Definition: Oversampling

Oversampling uses a sampling rate higher than the Nyquist rate: fs≫2fmax.

Benefits:

  • Relaxes anti-aliasing filter requirements
  • Spreads quantization noise over wider bandwidth
  • Enables noise shaping in sigma-delta converters

Definition: Decimation

Decimation by factor M reduces the sampling rate:

y[n]=x[nM]

Must be preceded by lowpass filtering to prevent aliasing.

Definition: Interpolation (Upsampling)

Interpolation by factor L increases the sampling rate:

y[n]={x[n/L]n=0,±L,±2L,…0otherwise

Must be followed by lowpass filtering to remove spectral images.