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Fundamentals of Digital Signal Processing

The document discusses fundamentals of digital signal processing including: 1) It describes sound modeling and classification of signals including signal, source, and abstract models. 2) It explains digital signals and how analog signals are converted to digital via sampling and analog-to-digital conversion. 3) It covers the discrete Fourier transform and how it is used to analyze digital signals in the frequency domain. The inverse discrete Fourier transform can then be used to reconstruct the signal.

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Tu Minh Hien
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0% found this document useful (0 votes)
121 views24 pages

Fundamentals of Digital Signal Processing

The document discusses fundamentals of digital signal processing including: 1) It describes sound modeling and classification of signals including signal, source, and abstract models. 2) It explains digital signals and how analog signals are converted to digital via sampling and analog-to-digital conversion. 3) It covers the discrete Fourier transform and how it is used to analyze digital signals in the frequency domain. The inverse discrete Fourier transform can then be used to reconstruct the signal.

Uploaded by

Tu Minh Hien
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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Fundamentals of digital signal

processing

1
Sound modeling

sound
symbols

aims
analysis
synthesis
processing

classification:
signal models
source models
abstract models

2
Digital signals
x(t) x(n) y(n) y(t)
analog 0.05 0.05 0.05 0.05

sampling
processing 0 0 0 0
reconstruction

-0.05 -0.05 -0.05 -0.05


0 500 0 10 20 0 10 20 0 500
t in µs ec n n t in µs e c

sampling interval T
sampling frequency fs= 1/T

3
Digital signals: time representations

8000 samples 0.5

100 samples

x(n)
0

line with dots -0.5


0 1000 2000 3000 4000 5000 6000 7000 8000
0.5

x(n)
0

-0.5
0 100 200 300 400 500 600 700 800 900 1000
0.05
vertical quantization x(n)

0
integer
e.g. -32768 .. 32767 -0.05
0 10 20 30 40 50 60 70 80 90 100
normalized n

e.g. -1 .. (1-Q)
Q = quantization step

4
Spectrum: analog vs. digital signal

sampling leads to a replication of the baseband spectrum

5
Spectrum: analog vs. digital signal

Sampling leads to a replication of the analog signal spectrum


Reconstruction of the analog signal:
low pass filtering the digital signal

6
Discrete Fourier Transform

Magnitude

Phase

7
Discrete Fourier Transform (example)

FFT with 16 points 1


Cos ine s ignal x(n)

0
cosine (16 points)

a)
-1
0 2 4 6 8 10 12 14 16
n
magnitude (16 points) 1
Ma gnitude s pe ctrum |X(k)|

normalization: 0.5

b)
0 dB for sinusoid ±1 0
0 2 4 6 8 10 12 14 16

magnitude 1
k
Ma gnitude s pe ctrum |X(f)|

(frequency points) 0.5

c)
kfs / N 0
0 0.5 1 1.5 2 2.5 3 3.5
step fs/N f in Hz x 10
4

20
magnitude dB vs. Hz Magnitude s pe ctrum |X(f)| in dB
0
|X(f)| in dB

-20

-40
0 0.5 1 1.5 2 2.5 3 3.5
f in Hz x 10
4

8
Inverse Discrete Fourier Transform (DFT)

if X(k) = X*(N-k)
then IDFT gives N discrete-time real values x(n)

9
Frequency resolution

Zero padding: to increase 8 s amples


10
8-point FFT

2
frequency resolution 8

1 6

|X(k)|
x(n)
4
0
2
-1
0
0 2 4 6 0 2 4 6

8 s amples + ze ro-pa dding 16-point FFT


10
2
8

1 6

|X(k)|
x(n)
4
0
2
-1
0
0 5 10 15 0 5 10 15
n k

10
Window functions

to reduce leakage:
weight audio samples by
a window

Hamming window
wH(n) = 0.54 – 0.46 cos(2 n/N)

Blackman window
wB(n) = 0.42 – 0.5cos(2 n/N) + 0.08(4 n/N)

11
Window

Reduction of the leakage effect by


window functions:
(a) the original signal,
(b) the Blackman window function
of length N =8,
(c) product x(n)w(n) with 0 n N-i,
(d) zero-padding applied to z(n)
w(n) up to length N = 16

The corresponding spectra are


shown on the right side.

12
Spectrograms

13
Waterfall representation

S ignal x(n)
1

0.5

0
x(n)

-0.5

-1
0 1000 2000 3000 4000 5000 6000 7000 8000
n
Waterfall Repres entation of S hort-time FFTs

0
Magnitude in dB

-50 0
2000
-100 4000
0 6000
5 10 15 20 n
f in Hz

14
Digital systems

15
Definitions

Unit impulse

Impulse reponse h(n) = output to a unit impulse


h(n) describes the digital sistem

Discrete convolution:
y(n)=x(n)*h(n)

16
Algorithms and signal graphs

Delay
e.g. y(n) = x(n-2)

Weighting factor
e.g. y(n) = a x(n)

Addition
e.g.
y(n) = a1 x(n) + a2 x(n)

17
Simple digital system

weighted sum over several input samples

18
Transforms

Frequency domain desciption of the digital system

Z transform

Discrete time Fourier


transform

Transfer function H(z):


Z transform of h(n)

Frequency response:
Discrete time Fourier
transform of h(n)

19
Causal and stable systems

Causality: a discrete-time system is causal


if the output signal y(n) = 0 for n <0 for a given input signal
u(n) = 0 for n <0.
This means that the system cannot react to an input before the
input is applied to the system

Stability: a digital system is stable if

stability implies that transfer function H(z) and frequency


response are related by

20
IIR systems

= system with infinte impulse response


e.g. second order IIR system

Difference equation

Transfer function

21
IIR systems

= system with infinte impulse response h(n)

Difference equation

Z transform of diff. eq.

Transfer function

22
FIR systems

= system with finite impulse response h(n)


e.g. second order FIR system

Difference equation

Z transform of diff. eq.

Transfer function

23
Fir example

computation of frequency response


(a ) Impuls e Re s pons e h(n) (b) Magnitude Res pons e |H(f)|
0.8
0.3
impulse response 0.2 0.6

magnitude resp. 0.1 0.4

|H(f)|
0
pole/zero plot 0.2
-0.1
0
phase resp. 0 2 4 0 10 20 30 40
n f in kHz
(c ) P ole/Zero plot (d) P ha s e Res pons e H(f)
0

1
-0.5

Im(z) 4

H(f)/
0 -1

-1.5
-1

-2
-1 0 1 2 0 10 20 30 40
Re (z) f in kHz

24

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