Chapter 3

Sampling, Reconstruction, and DT

Filtering
1. Sampling (continuous-to-discrete time conversion)
2. Reconstruction (discrete-to-continuous time conversion)
3. Discrete-time filtering of continuous-time signals
Digital processing of analog signals
Typical system x c (t) x[n] y[n]
Digital Signal yc(t)
ADC DAC
Processor

Analog-to-digital converter (ADC)

► Performs filtering, sampling, and quantization
► Sampling rate may be of tens of kHz (audio processing), or it may be of tens of GHz
(optical communications)
Digital signal processor
► Performs some operation e.g., filtering, FFT, etc
► May be implemented on PCs with 64-bit floating-point precision, or on ASICs with limited
arithmetic precision (e.g., 6 bits).
Digital-to-analog converter (DAC)
► Performs quantization and reconstruction (filtering)
► Sampling rate could be similar to ADC 4/47
Analog-to-digital conversion
In practice
Example of successive-approximation analog-to-digital converter (SA-ADC)
Analog-to-digital conversion

In this class
We’ll model the ADC as an ideal continuous-to-discrete (C-to-D) time converter.

x c (t) x[n] = xc (nT )

C-to-D
X c (jΩ) Converter X (ej ω ), X (ej ΩT )

Notation:
X c (jΩ) denotes the Fourier transform of the continuous-time signal x c (t), where Ω is the
continuous-time frequency.
X(e jω ) denotes the discrete-time Fourier transform of a discrete-time signal x[n], where ω is
the normalized frequency.
Continuous-to-discrete time conversion
The C-to-D converter simply samples the continuous-time signal every T seconds, where T is
the sampling period.
x c (t)

3 x[n] = x c (nT )

T 2T 3T . . . t

Question: How are x c (t) and x[n] related in the frequency domain? That is, how to obtain the
discrete-time Fourier transform X(e jω ) from the continuous-time Fourier transform X c (jΩ)? 7/47
Continuous-to-discrete time conversion
Impulse sampling interpretation:
We can think of the C-to-D converter as multiplication by an impulse train, followed by an
impulse-to-sequence converter.
This representation is purely for mathematical convenience.
Impulse sampling example
x c (t)

s(t)

t
x s (t) = x c (t) · s(t)

x[n]

n
Continuous-to-discrete time conversion
Question: How to obtain the discrete-time Fourier transform X(e jω ) from the
continuous-time Fourier transform X c (jΩ)?

We’ll calculate the continuous-time (CT) Fourier transform of x s (t) in two different ways.
1 X (j Ω) ∗S(j Ω). Then, we’ll calculate X (j Ω) = F { x (t)}
First, we’ll calculate X s(j Ω) = 2π c s s .
We’ll use these two equations of X s (jΩ) to obtain an equation for X(e jω ), the DTFT of x[n].
Starting with X s(jΩ) = Xc(jΩ) ∗S(jΩ), recall that the Fourier Transform of the
impulse train is given by

Now we can calculate X s (jΩ):

1
X s (j Ω) = X (j Ω) ∗S(j Ω)
2π c

(1)

Note that X s (jΩ) is equal to X c (jΩ) scaled by 1/T and repeated every 2π/T, which we
define as the sampling frequency Ωs ≡ 2π/T 10/47
Now let’s calculate X s (jΩ) = F{x s (t)}, where x s (t) = x c (t) ·s(t):

X s (jΩ) is equal to X(e jω ) evaluated at ω = ΩT , or equivalently X(e jω ) is equal to X s (jΩ)

evaluated at Ω = ω/T. 12/47
 Substituting (1) in (2):

where we used the relation ω = ΩT.

► T is the sampling period, and Ωs = 2π is the sampling frequency
T
► This equation shows that in discrete time (ω = ΩT) replicas of the original spectrum appear
with period 2π

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Graphically
X c ( jΩ)
1

−ΩN ΩN Ω
S(jΩ)
2π
T

−Ωs Ωs Ω
1
X s(jΩ) = 2π
X c(j Ω) ∗ S(j Ω)
1/T

−Ωs −ΩN ΩN Ωs Ω
X (ej ω ) = X (j Ω)
1/T

− 2π − ΩN T ΩN T 2π ω = ΩT
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Replicas of the original spectrum appear with period 2π
Oversampling
► A signal is band limited if X c (jΩ) = 0 for |Ω| > Ω N . In this case, the signal has
maximum frequency Ω N and bandwidth 2ΩN
► Sampling at Ωs > 2ΩN is called oversampling
► Oversampling leads to gaps between the spectrum replicas
Xc(jΩ)

Ωs Ωs
− 2 2

−Ωs −ΩN ΩN Ωs Ω
X (ej ω )
Ωs > 2Ω N
1/T

−π π
− 2π − ΩN T ΩN T 2π ω = ΩT
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Nyquist sampling
► Sampling at Ωs = 2ΩN is called Nyquist sampling
► Note that if Ωs is any smaller than 2ΩN , there will be overlapping of the spectrum replicas
X c (jΩ)

Ωs Ωs
− 2 2

−Ωs −ΩN ΩN Ωs Ω
X (ej ω ) Ωs = 2ΩN
1/T

−π π
− 2π − Ω N T ΩN T 2π ω = ΩT
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Undersampling
► Undersampling occurs when Ωs < 2ΩN
► In this case, the spectrum replicas overlap
► The overlapping causes aliasing distortion
X c ( jΩ)

−ΩN ΩN Ω
X (ej ω )
Ωs < 2Ω N
1/T

− 2π −π π 2π ω = ΩT 17/47
Aliasing: time domain
Samples taken at frequency Ωs = 2π form an alias signal of frequency 0.5π, but original signal
had frequency 1.5π

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Aliasing: frequency domain
Same example, but now in the frequency domain

X(jω)

− 1.5π 1.5π Ω
X (ej ω )

− 2π −π 0.5π π 1.5π 2π ω = ΩT

Blue components correspond to spectrum replica centered at 2π, while red components correspond
to spectrum replica centered at −2π.
The final spectrum corresponds to cos(0.5πn) 19/47
Digital-to-analog conversion
In practice
Example of digital-to-analog converter (DAC)

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Digital-to-analog conversion

In this class
We’ll model the ADC as an ideal discrete-to-continuous (D-to-C) time converter.

x[n] D-to-C x c (t)

X (ej ω ),X (ej ΩT ) Converter X (j Ω)

In essence, a D-to-C converter performs interpolation.

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Discrete-to-continuous time conversion
For mathematical convenience we can model the D-to-C as

x[n] Discrete-time
Lowpass filter x r (t)
sequenceto
hr (t) H r (j Ω)
impulse train

D-to-C converter

∞
xr (t) = � x[n]hr (t − nT) (D-to-C converter)
n=−∞

Important questions
1. How close to the original signal is x r (t)?
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2. What lowpass filter H r (jΩ) will lead to the best performance?
Reconstruction: time domain x[n]

x s (t) = ∑ x[n]δ(t − nT )

x r (t) = x s (t) ∗ hr (t)

t
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Reconstruction: frequency domain
X (ej ω )

1/T

− 2π − ΩN T ΩN T 2π ω = ΩT
X s (j Ω) = X (ej ΩT )

1/T

−Ωs −ΩN ΩN Ωs Ω
X r (j Ω) = H r (j Ω)X s (j Ω)

1 Which lowpass filter

would produce this result?

− ΩN ΩN Ω 27/47
Reconstruction: frequency domain
X (ej ω )

1/T

− 2π − ΩN T ΩN T 2π ω = ΩT
X s (j Ω) = X (ej ΩT )

1/T

−Ωs −ΩN ΩN Ωs Ω
X r (j Ω) = H r (j Ω)X s (j Ω)

− ΩN ΩN Ω 27/47
Shannon-Nyquist sampling theorem

Shannon-Nyquist sampling theorem

A band-limited signal with highest frequency Ω N can be perfectly reconstructed from
samples taken with sampling frequency Ωs = 2πT > 2ΩN .

X r (j Ω) = Hr (j Ω)X (ej ΩT ) = X c (j Ω)

► Sampling above the Nyquist frequency (2ΩN ) avoids aliasing

► In practice, it is common to use an anti-aliasing filter to minimize aliasing when the analog
signal is not band-limited.
► Perfect reconstruction is achieved if H r (jΩ) is the ideal lowpass filter. In other words, the ideal
lowpass filter (or sinc function in time domain) is the perfect interpolator for
band-limited signals.
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 Ideal lowpass filter
Time domain Frequency domain
π
sin πTt t T, |Ω| ≤
hlp f (t) = π = sinc ( ) Hlp f (j Ω) = T
π
T
t T 0, |Ω| > T

hlpf (t) Hlpf (j Ω)

1 T

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−3T −2T −T T 2T 3T t Ω
−π/T π/T
Example of reconstruction with an ideal lowpass filter
∞ ∞
x r (t) = � x[n]hr (t − nT) = � x[n]sinc(t − nT) (reconstruction)
n= − ∞ n= − ∞

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Original continuous-time signal
Example of reconstruction with an ideal lowpass filter
∞ ∞
x r (t) = � x[n]hr (t − nT) = � x[n]sinc(t − nT) (reconstruction)
n= − ∞ n= − ∞

Samples from original continuous-time signal

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Example of reconstruction with an ideal lowpass filter
∞ ∞
x r (t) = � x[n]hr (t − nT) = � x[n]sinc(t − nT) (reconstruction)
n= − ∞ n= − ∞

At the nth sample, we have the sinc function x[n]sinc(t − nT)

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Example of reconstruction with an ideal lowpass filter
∞ ∞
x r (t) = � x[n]hr (t − nT) = � x[n]sinc(t − nT) (reconstruction)
n= − ∞ n= − ∞

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The sum of all sincs results in the perfectly reconstructed signal.
Practical reconstruction

Problem
The ideal lowpass filter is not feasible, as it is non-causal and requires infinitely many samples.

Common reconstruction filters

1. Zero-order hold (square pulse)
2. Linear interpolation (triangular pulse)
3. Cubic spline interpolation

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