Basics on Digital Signal Processing

Introduction

Vassilis Anastassopoulos
Electronics Laboratory, Physics Department,
University of Patras
 Outline of the Course

1. Introduction (sampling – quantization)

2. Signals and Systems
3. Z-Transform
4. The Discreet and the Fast Fourier Transform
5. Linear Filter Design
6. Noise
7. Median Filters

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 Analog & digital signals
Analog Digital
Continuous function V Discrete function Vk of
of continuous variable t
Sampled discrete sampling
(time, space etc) : V(t). Signal variable tk, with k =
integer: Vk = V(tk).

0.3 0.3
0.2 0.2
Voltage [V]

Voltage [V]
0.1 0.1
0 0
-0.1 -0.1 ts ts
-0.2 -0.2
0 2 4 6 8 10 0 2 4 6 8 10
time [ms] sampling time, tk [ms]
Uniform (periodic) sampling.
Sampling frequency fS = 1/ tS

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Analog & digital systems

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 Digital vs analog processing
Digital Signal Processing (DSPing)
Advantages Limitations

• More flexible. • A/D & signal processors speed:

wide-band signals still difficult to
• Often easier system upgrade. treat (real-time systems).
• Data easily stored -memory. • Finite word-length effect.
• Better control over accuracy
requirements.
• Reproducibility.
• Linear phase
• No drift with time and
temperature

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 DSPing: aim & tools
• Predicting a system’s output.
Applications • Implementing a certain processing task.
• Studying a certain signal.

• General purpose processors (GPP), -controllers.

Hardware • Digital Signal Processors (DSP). Fast real-time
Faster DSPing
• Programmable logic ( PLD, FPGA ).

• Programming languages: Pascal, C / C++ ...

Software • “High level” languages: Matlab, Mathcad, Mathematica…
• Dedicated tools (ex: filter design s/w packages).

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Related areas

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Applications

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Important digital signals
Unit Impulse or Unit Sample.
δ(nTs) δ[(n-3)Τs]
The most important signal for
two reasons
nΤs past δ(n)=1 for n=0

u(nTs)
Unit Step u(n)=1 for n0

nΤs past
δ(n)=u(n)-u(n-1)

r(nTs)

Unit Ramp r(n)=nu(n)

nΤs past

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 Digital system example
V

DOMAIN
ANALOG
General scheme Filter
ms
Filter
Antialiasing
V
Antialiasing
Sometimes steps missing
ms
- Filter + A/D A
A/D
A/D

DOMAIN
DIGITAL
(ex: economics);
k
- D/A + filter Digital
Processing
Digital
A
(ex: digital output wanted).
k Processing
V
D/A

DOMAIN
ANALOG
Topics of this ms
Filter
V
lecture. Reconstruction

ms

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Digital system implementation
KEY DECISION POINTS:
ANALOG INPUT
Analysis bandwidth, Dynamic range

Antialiasing
Filter • Pass / stop bands.
1
• Sampling rate.
A/D
• No. of bits. Parameters. 2
Digital
Processing • Digital format. 3
What to use for processing?

DIGITAL OUTPUT

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AD/DA Conversion – General Scheme

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AD Conversion - Details

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Sampling

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1 Sampling
How fast must we sample a continuous
signal to preserve its info content?

Ex: train wheels in a movie.

25 frames (=samples) per second.

Train starts wheels ‘go’ clockwise.

Train accelerates wheels ‘go’ counter-clockwise.

Why?
Frequency misidentification due to low sampling frequency.

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 Rotating Disk

How fast do we have to instantly

stare at the disk if it rotates
with frequency 0.5 Hz?

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 1 The sampling theorem
A signal s(t) with maximum frequency fMAX can be
Theo* recovered if sampled at frequency f > 2 f
S MAX .

* Multiple proposers: Whittaker(s), Nyquist, Shannon, Kotel’nikov.

Naming gets
confusing ! Nyquist frequency (rate) fN = 2 fMAX or fMAX or fS,MIN or fS,MIN/2

Example
s(t)  3  cos(50π t)  10  sin(300π t)  cos(100π t) Condition on fS?
F1 F2 F3

F1=25 Hz, F2 = 150 Hz, F3 = 50 Hz fS > 300 Hz

fMAX

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Sampling and Spectrum

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1 Sampling low-pass signals
Continuous spectrum
(a) (a) Band-limited signal:
frequencies in [-B, B] (fMAX = B).

-B 0 B f

(b) Discrete spectrum

No aliasing
(b) Time sampling frequency
repetition.
fS > 2 B no aliasing.
-B 0 B fS/2 f

Discrete spectrum
(c) Aliasing & corruption
(c) fS 2B aliasing !

Aliasing: signal ambiguity

0 fS/2 f in frequency domain

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 1 Antialiasing filter
(a) Signal of interest (a),(b) Out-of-band noise can aliase
Out of band
Out of band into band of interest. Filter it before!
noise
noise

-B 0 B f
(c) Antialiasing filter
(b) Passband: depends on bandwidth of
interest.

Attenuation AMIN : depends on

-B 0 B fS/2 • ADC resolution ( number of bits N).
(c) f
AMIN, dB ~ 6.02 N + 1.76
• Out-of-band noise magnitude.

Other parameters: ripple, stopband

frequency...

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 1 Under-sampling
Using spectral replications to reduce Bandpass signal
B
sampling frequency fS req’ments. centered on fC

0 fC
2  fC  B 2  fC  B
 fS  f
m 1 m

m , selected so that fS > 2B

-fS 0 fS 2fS
f fC
Example
fC = 20 MHz, B = 5MHz Advantages
Without under-sampling fS > 40 MHz.  Slower ADCs / electronics
With under-sampling fS = 22.5 MHz (m=1);
needed.

= 17.5 MHz (m=2); = 11.66 MHz (m=3).  Simpler antialiasing filters.

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 Quantization and Coding
N Quantization Levels

Quantization Noise

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 2 SNR of ideal ADC
 RMS input  Assumptions
SNRideal  20  log10   (1)
 RMS(e q )   Ideal ADC: only quantisation error eq
 
(p(e) constant, no stuck bits…)
Also called SQNR
 eq uncorrelated with signal.
(signal-to-quantisation-noise ratio)
 ADC performance constant in time.

T 2
RMS input 
1  VFSR
 
 V
 sinωt  dt  FSR
Input(t) = ½ VFSR sin( t).
T  2  2 2
0
p(e)
quantisation error probability density

q/2
RMS(e q )    
eq2  p eq deq 
q

VFSR
12 2N  12
1
q
-q/2

(sampling frequency fS = 2 fMAX) q q eq

2 2
Error value

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 2 SNR of ideal ADC - 2
Substituting in (1) : SNRideal  6.02  N  1.76 [dB] (2)

One additional bit SNR increased by 6 dB

Real SNR lower because:

- Real signals have noise.
- Forcing input to full scale unwise.
- Real ADCs have additional noise (aperture jitter, non-linearities etc).

Actually (2) needs correction factor depending on ratio between sampling freq
& Nyquist freq. Processing gain due to oversampling.

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Coding - Conventional

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Coding – Flash AD

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DAC process

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 Oversampling – Noise shaping
PSD Nyquist Sampler

f
fb fN (a) The oversampling process takes apart
Oversampling OSR=4
the images of the signal band.

f
(b) fs=4fN
PSD
Quantization noise in When the sampling rate increases (4
Signal Nyquist converters
times) the quantization noise spreads
Quantization noise in over a larger region. The quantization
Oversampling converters
noise power in the signal band is 4 times
0 fN/2
smaller.
fs/2

Quantization noise
PSD
Signal
Quantization noise Oversampling and noise Spectrum at the output of a noise
Nyquist converters shaping converters
shaping quantizer loop compared to
Quantization noise
Oversampling converters those obtained from Nyquist and
Oversampling converters.
0 FN/2 frequency Fs/2

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 Digital Systems
A discreet-time system is a device or algorithm
that operates on an input sequence according to
some computational procedure

It may be
•A general purpose computer
•A microprocessor
•dedicated hardware
•A combination of all these

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Linear, Time Invariant Systems
System Properties N
• linear
•Time Invariant
y ( n )   ak x ( n  k )
•Stable k 0
Convolution
•Causal

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Linear Systems - Convolution

5+7-1=11 terms

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Linear Systems - Convolution

5+7-1=11 terms

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General Linear Structure

M L
y (n)   ak x(n  k )   bk y (n  k )
k 0 k 1

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Simple Examples

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 Linearity – Superposition – Frequency Preservation

Principle of Superposition
x1(n) y1(n)
H
ax1(n)+bx2(n) ay1(n)+by2(n)
H
x2(n) y2n)
H

Principle of Superposition  Frequency Preservation

x1(n) x12(n)
x2

Non-linear
x1(n)+x2(n) x12(n)+x22(n)+2 x1(n) x2(n)
x2
x2(n) x22(n)
2
x

If y(n)=x2(n) then for x(n)=sin(nω) y(n)=sin2(nω)=0.5+0.5cos(2nω)

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 The END

Have a nice Weekend

Back on Tuesday

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