ICE 3018

2. Discrete Signals
ICE3018: Digital Signal Processing & Design

School of Information & Communication Engineering

Inha University

Image credit: SINTEF

2. Discrete Signals
Things to learn
1. Time shift and time reversal for discrete signal
2. Even symmetry and odd symmetry
3. Classification of a discrete signal based on energy and power
4. Basics of interpolation and decimation
5. Definitions of standard signal forms
6. Digital frequency of a sinusoid or complex harmonics
7. The common period of a sum of sinusoids or complex
harmonics.
8. Sampling theorem and sampling rate
9. Aliasing

2
2.1 Discrete Signals
Definition:
A sampled or discrete signal x[n] is just an ordered
sequence of values corresponding to the integer index n.

x(t) : analog signal (t: continuous time variable)

x[n]: discrete signal (n: integer)
n=0
⇓ ⇓
40

x[n] = {1, 2, 4, 8} vs. y[n] = {8, 4, 2, 1}

30

20

10

(ML) x=[1, 2, 4, 8]; n=[0, 1, 2, 3]; 0

-10

-20

-30

-40 x[0], x[1], x[2], …, x[n], …, x[20]

3
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
2.1 Discrete Signals
 Infinite-length signals
 Left-sided if x[n] = 0 for n>N0
Where N0 is a certain integer.
 Right-sided if x[n] = 0 for n<N0
 Causal if x[n] = 0 for n<0  Positive-time sequence
 Anti-causal if x[n] = 0 for n>0  Negative-time sequence

N0) N0) 0) > 0)

N0 N0

4
2.1 Discrete Signals
 Signal Measures
 Summation (equivalent to integration for continuous signal)
∞ ∞
Discrete sum S D = ∑ x[n],
n = −∞
Absolute sum S A = ∑ x[n]
n = −∞
n
Cummulative sum SC [n] = ∑ x[k ] (also Running sum)
k = −∞
 x[n]is absolutely summable if its SA is finite.
 Instantaneous power p[n] = x[n]
2

 Signal energy of non-periodic signal

∞ ∞ 2

E= ∑ p[m] = ∑ x[m]
m = −∞ m = −∞

 x[n] is called an Energy signal if E is finite

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2.1 Discrete Signals
 Periodic signals
x[n] = x[n ± kN], k=0, 1, 2, …
where N is the smallest number of samples that repeat and is
always an integer.
1 N −1
Average value : xav =
N
∑ x[m]
m =0

1 N −1

∑ x[m]
2
Average power : P =
N m =0

 For non-periodic signals

1 L
Average value : xav = lim ∑ x[m]
L →∞ 2 L + 1
m=− L

1 L

∑ x[m]
2
Average power : P = lim
L →∞ 2 L + 1
m=− L

 x[n] is called a Power signal if its P is finite.

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2.2 Operations on Discrete Signals
 Time shift
 Delayed when α > 0
y[n] = x[n − α ] 
Advanced when α < 0
Letting m = n − α , n = m + α . Hence, y[m + α ] = x[m]

 Time reversal (Mirror image)

 Replace n by (-n) y[n] = x[− n]
 Folding: reversal around n=0

 Shift-and-reversal
x[n] → [delay (shift right) by α ] → x[n − α ] → [reversal] → x[− n − α ]
x[n] → [reversal] → x[− n] → [advance (shift left) by α ] → x[− (n + α )]

1
7
2.2 Operations on Discrete Signals
 Symmetry
 Even symmetric xe [n] = xe [− n]
 Odd symmetric (Anti-symmetric) xo [n] = − xo [− n]

8
2.2 Operations on Discrete Signals
 Decomposition of Arbitrary signal
x[n] = xe [n] + xo [n]

1
where xe [n] = {x(n ) + x(− n )} and xo [n] = 1 {x(n ) − x(− n )}
2 2
 Any signal can be expressed as the sum of an even symmetric
part and an odd symmetric part
 Even symmetry and odd symmetry are mutually exclusive.

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2.2 Operations on Discrete Signals

 Examples of even functions?

 Examples of odd functions?

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2.3 Decimation and Interpolation
 Decimation
 Reducing the signal length by discarding signal samples
 Decimation by 2: y[n] = x[2n] = {x[0], x[2], x[4], …}
 Length of y[n] is reduced by a factor of 2
 Decimation by N: y[n] = x[Nn] = {x[0], x[N], x[2N], x[3N], …}
 Length of y[n] is reduced by a factor of N.

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2.3 Decimation and Interpolation
 Interpolation
 Increasing the signal length by inserting signal samples
 Interpolation by 2: y[n] = x[n/2] = {x[0], x[1/2], x[2/2], x[3/2], …}
 Length of y[n] is increased by a factor of 2
 Interpolation by N: y[n] = x[n/N]
 Need to insert (N-1) samples between x[n] and x[n+1]

 For a discrete signal x[n], how to decide new samples?

 Zero interpolation (Up-sampling): all new samples are zero
 Step interpolation: equal to the previous sample value
 Linear interpolation: average of adjacent samples

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2.3 Decimation and Interpolation
 Are decimation and interpolation inverse operations?

 Decimation is indeed the inverse of interpolation, but the

converse is not necessarily true.
 When both operations need to be performed in
succession, interpolate first!

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Application of Decimation and Interpolation

 M
 Fractional Delays y[n] = x n −  where M, N are integers.
 N

 M  Nn − M 
Generating y[n] = x n −  = x   from x[n]
 N  N 

n n − M 
x[n] ⇒ [Interpolation by N ] ⇒ x ↑   ⇒ [Delay by M ] ⇒ x 
N   N 
 Nn − M 
⇒ [Decimation by N ] ⇒ x   = y[n]
 N 

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2.4 Standard Discrete Signals
 Unit impulse (or Unit sample)  Dirac delta function
0 , n ≠ 0
δ [n] = 
 Unit step 1, n = 0
0 , n < 0
u[n] = 
1, n ≥ 0
 Unit ramp
0 , n < 0
r [n] = nu[n] = 
n , n ≥ 0

δ [n] = u[n] − u[n − 1]

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2.4 Standard Discrete Signals
 Properties of Discrete Impulse
 Product of x[n] with δ[n-k]: x[n]δ [n − k ] = x[k ]δ [n − k ]
 Sifting property
∞

∑ x[n]δ [n − k ] = x[k ]
n = −∞
Impulse with the
magnitude x[k]

x[n] δ[n-3]
1
1
2

x =
k=3 k=3

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2.4 Standard Discrete Signals
 Signal Representation by Impulses
 Expressing x[n] as a sum of shifted impulses δ[n-k]
x[n]
∞
x[n] = ∑ x[k ]δ [n − k ]
k = −∞

∞
x[-1]×δ[n+1]
Ex: 𝑢𝑢 𝑛𝑛 = � 𝛿𝛿 𝑛𝑛 − 𝑘𝑘 and
𝑘𝑘=0
x[0]×δ[n]
𝑟𝑟 𝑛𝑛 = ∑∞
𝑘𝑘=0 𝑘𝑘 ⋅ 𝛿𝛿 𝑛𝑛 − 𝑘𝑘

x[1]×δ[n-1]
 As a cumulative sum (or running sum)
n n −1
u[n] = ∑ δ [k ] and r[n] = ∑ u[k ]
k = −∞ k = −∞
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2.4 Standard Discrete Signals
 Discrete Pulse Signals
 n  1, n≤N
• Discrete rectangular pulse : rect  =
 2 N  0 , elsewhere
 n
n
  
• Discrete triangular pulse : tri  = 1 − N , n≤N
 N   0, elsewhere

 Both signal has (2N+1) samples.

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2.4 Standard Discrete Signals

 Discrete Sinc Function

 nπ 
sin 
n  N  , and sinc (0 ) = 1
sinc   = >> x = linspace(-5,5);
N  nπ 
  >> y = sinc(x);
 N  >> stem(x,y)

sinc(n/N) = 0 at n=kN, k=±1, ±2, …

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Transform of Sinc function

Time Freq
Transformation
domain Domain

Box-car Sinc
Fourier Transform

Sinc Inverse Fourier Box-car
Transform

F.T.

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Continuous Sinc Function

 Rectangular pulse signal: Window (Truncation) function

Forward FT

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Continuous Sinc Function
Transfer function of Ideal
 Bandlimited signal: Low-pass Filter

Inverse FT

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2.4 Standard Discrete Signals
 Discrete Exponentials
 Real-valued exponential signal
x[n] = a nu[n] - decaying for a < 1

 Complex-valued exponential signal

x[n] = (a + jb ) u[n] = re
n
( ) u[n]
jω o n
r : attenuation when r < 1

= r n e jnωo u[n] = r n [cos (nωo ) + jsin(nωo )]u[n] ωo : frequency
 Sinusoidal signal

𝐴𝐴: amplitude
𝑥𝑥 𝑛𝑛 = Acos 𝜔𝜔o 𝑛𝑛 + 𝜃𝜃 �𝜔𝜔o : frequency
𝜃𝜃: phase

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2.5 Discrete-time Harmonics and Sinusoids

 Analog freq. vs. Digital freq.

x(t ) = A cos(2πft + θ ) or A cos(ωt + θ )
t  n⋅ts  n
Sampling
t by ts = 1/ S (S =samples/sec)
0 ts 2ts 3ts
𝑓𝑓
𝑥𝑥 𝑛𝑛 = 𝐴𝐴cos 2𝜋𝜋𝜋𝜋𝜋𝜋𝑡𝑡𝑠𝑠 + 𝜃𝜃 = 𝐴𝐴cos 2𝜋𝜋𝜋𝜋 + 𝜃𝜃 = 𝐴𝐴cos 2𝜋𝜋𝜋𝜋𝜋𝜋 + 𝜃𝜃
S

Analog: 𝑡𝑡 sec → 𝑓𝑓 cycles/sec → 𝜔𝜔 = 2𝜋𝜋𝜋𝜋 rad/sec

𝑓𝑓
Digital: 𝑛𝑛 ⋅ → 𝐹𝐹 = 𝑓𝑓ts = cycles/sample → Ω = 2𝜋𝜋𝜋𝜋 rad/sample
𝑆𝑆

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Example: Digital frequency
𝑓𝑓 = 100 Hz

8 samples/cycle
𝑡𝑡𝑠𝑠 =?
𝐹𝐹 = 𝑓𝑓𝑡𝑡𝑠𝑠 =?

𝑓𝑓 = 100 Hz

1 sample/cycle
𝑡𝑡𝑠𝑠 =?
𝐹𝐹 = 𝑓𝑓𝑡𝑡𝑠𝑠 =?

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Analog vs. Digital Signals

t
0 ts 2ts 3ts

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Sampling – How fast?
 Consider a pure sine wave of frequency 100Hz:
𝑥𝑥 𝑡𝑡 = A𝑠𝑠𝑠𝑠𝑠𝑠 2π𝑓𝑓𝑓𝑓 + 𝜃𝜃

 In order to ensure that we retain all of the information in

the signal, what sampling rate should be used? (no
quantization)

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Sampling – Too fast ?
(1) Sampling at fs = 800Hz, i.e., 8 samples per period:

“Reasonable”

(2) Sampling at fs = 3,000Hz, i.e., 30 samples per period:

“Redundant”

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Sampling – Too slow?
(3) Sampling at fs = 100Hz, i.e., 1 sample per period:

Signal
interpreted as
DC

(4) Sampling at fs = 80Hz, i.e., 1 sample per 1.25 period:

Most signal
features are
missed

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2.6 The Sampling Theorem
 Sampling rate or sampling frequency, fs (Hz) : the speed
at which an ADC generates the samples.

 Sampling interval or sampling period, ts (sec): the time

between samples.
1
ts =
fs

What is the appropriate sampling frequency or

sampling period not to lose signal information?
Sampling theorem

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Signal Frequency Range Terminology

 Baseband signal:
 the lowest signal frequency present is around 0 Hz :

fb : bandwidth

 Bandlimited signal:
 for all frequencies in the signal fl < f < fh :

31
Illustration of Sampling
Time domain Frequency domain

𝑥𝑥(𝑡𝑡) 𝑋𝑋(𝑓𝑓)

𝑡𝑡 𝑓𝑓
0 −𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚 0 𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚

𝑠𝑠(𝑡𝑡) 𝑆𝑆(𝑓𝑓)
𝑡𝑡𝑠𝑠 1/𝑡𝑡𝑠𝑠 = 𝑓𝑓𝑠𝑠

⋯ ⋯ ⋯ ⋯
𝑡𝑡 𝑓𝑓
0 0

𝑥𝑥(𝑡𝑡) × 𝑠𝑠(𝑡𝑡) 𝑋𝑋 𝑡𝑡 ∗ 𝑆𝑆(𝑓𝑓)

1/𝑡𝑡𝑠𝑠 = 𝑓𝑓𝑠𝑠

⋯ ⋯ ⋯ ⋯
𝑡𝑡 𝑓𝑓
0 −𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚 0 𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚 32
Illustration of Sampling

𝑋𝑋 𝑡𝑡 ∗ 𝑆𝑆 𝑓𝑓

⋯ ⋯

𝑓𝑓
−1/𝑡𝑡𝑠𝑠 = −𝑓𝑓𝑠𝑠 −𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚 0 𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚 1/𝑡𝑡𝑠𝑠 = 𝑓𝑓𝑠𝑠

𝑓𝑓𝑠𝑠 − 𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚
Appropriate sampling frequency ?

𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚 < 𝑓𝑓𝑠𝑠 − 𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚

𝑓𝑓𝑠𝑠 < 2𝑓𝑓𝑚𝑚𝑚𝑚𝑚𝑚

33
“Suitable” Sampling Rate
 Nyquist’s Sampling Theorem :

f s ≥ 2 × f max
where fmax is the maximum frequency component of a
baseband, bandlimited signal.
fmax = 100Hz

36
Nyquist Sampling Rate
 If we require to sample the signal and retain all
information, then the sampling rate, fs must be chosen as:

fs > fN = 2fb

 This frequency (2fb) is referred to as the Nyquist sampling

rate.

37
Aliasing
 What if fs < 2fb ?
𝑋𝑋 𝑡𝑡 ∗ 𝑆𝑆 𝑓𝑓

⋯ ⋯

𝑓𝑓
−𝑓𝑓𝑠𝑠 − 𝑓𝑓𝑏𝑏 −𝑓𝑓𝑠𝑠 −𝑓𝑓𝑠𝑠 + 𝑓𝑓𝑏𝑏 −𝑓𝑓𝑏𝑏 0 𝑓𝑓𝑏𝑏 𝑓𝑓𝑠𝑠 − 𝑓𝑓𝑏𝑏 𝑓𝑓𝑠𝑠 𝑓𝑓𝑠𝑠 + 𝑓𝑓𝑏𝑏

⋯ ⋯

𝑓𝑓
−𝑓𝑓𝑠𝑠 −𝑓𝑓𝑏𝑏 0 𝑓𝑓𝑏𝑏 𝑓𝑓𝑠𝑠
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Aliasing
 When a (baseband) signal is sampled at a frequency
below the Nyquist rate, then we “lose” the signal
frequency information and aliasing is said to have
occurred.
fb = 100Hz
fs = 80Hz
fa = 20Hz

fb = 9000Hz
fs = 10000Hz
fa = 1000Hz

39
Aliased Spectra

40
Quantization

• If the ADC has finite precision due to a limited number of discrete

levels, then there may be a small error associated with each sample.
𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷𝐷 𝑟𝑟𝑟𝑟𝑟𝑟𝑟𝑟𝑟𝑟
𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄 𝐿𝐿𝐿𝐿𝐿𝐿𝐿𝐿𝐿𝐿 𝑄𝑄 = (n: #of bits)
2𝑛𝑛 − 1
𝑄𝑄
𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄𝑄 𝐸𝐸𝐸𝐸𝐸𝐸𝐸𝐸𝐸𝐸 Δ𝑞𝑞 =
2
46
 Reproducing an Analog Signal
 Using a DAC at an appropriate sampling rate, we can reproduce an
analog signal:

 The output is a staircase by the zero order hold.

 A zero order circuit is a form of capacitive element whereby the input
voltage is held constant for one sampling period.
 This artifact can however be removed by a reconstruction filter.
47
First Order Hold
 Alternatively, a first order hold could be attempted in the
DAC. The voltage between two discrete samples is
approximated by a straight line.

 Implementation of a first order circuit is not trivial.

48
Anti-aliasing Filter
 Prior to ADC, all frequencies above fs/2 must be blocked
or they will be interpreted as lower frequencies, i.e.,
aliasing:

 Anti-alias filter is analog (box-car filter), cutting off just

after fs/2.
49
Signal Aliasing
 Without an anti-alias filter, frequencies above fs/2 will “alias” to
lower frequencies and interfere with the baseband signal.
 Music signal sampled at fs=8kHz with NO anti-alias filter used :

 Frequencies of the music and speech above 4kHz alias down to the
0-4kHz baseband and manifest as (correlated) noise.

50
Reconstruction Filter
 Analog reconstruction filter at the output of a DAC
removes the baseband image high frequencies present
in the signal:

Staircase output
Analog voltage
from DAC

51
Real World Sampling Frequency
 To sample a real-world signal, an appropriate sampling rate is
chosen according to quantitative and qualitative constraints.
 Quantitative constraints will be specified by a domain expert, usually
in terms of the Nyquist rate.
 Qualitative constraints usually refer to acceptable quality.
 When digital signal sampled at fs is reconstructed, no frequency
components above fs /2 exist.

 CD Audio: 44,100Hz
 Professional Digital Audio: 48,000Hz
 Teleconferencing: 16,000Hz
 Telephone Speech: 8,000Hz
 Biomedical: 1,000Hz
 Control Systems: 100Hz

52
Real World Bit Rates
 To sample a real-world signal, a suitable data wordlength, is also
chosen according to quantitative and qualitative constraints.
 Quantitative constraints will be specified by a domain expert, usually
in terms of required SNR.
 Qualitative constraints usually refer to acceptable quality.
 When digital signal is sampled, quantized, and then reproduced, the
signal will have quantization noise present.

 CD Audio: 16 bits
 Professional Digital Audio: 24 bits
 Teleconferencing: 12-16 bits
 Telephone Speech: 8-13 bits
 Biomedical: 16-24 bits
 Control Systems: 4-10 bits

53
Summary of Ch 2.

 Basic terms and definitions for discrete signals

 Fundamental concepts and operations for
discrete signals
 Decimation and interpolation
 Analog freq vs. Digital freq
 Sampling theorem
 Aliasing

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