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continous-time signals

4.4 digital processing of analog signals

4.1 periodic sampling

1.ideal sample

x[n] xc (t ) |t nT xc (nT ) Tsample period

fs=1/T:sample rate

s=2/T:sample rate

Figure 4.1 ideal continous-time-to-discrete-time(C/D)converter

time normalization

tt/T=n

(t nT )

n

Figure 4.3

frequency spectrum

change of ideal sample

X ( j( k ))

1

X s ( j)

s N N T

c s

k

No aliasing

s N N

aliasing

X (e j ) X s ( j) | / T

s / 2

2

X ( j( k 2 ) / T )

1

aliasing frequency T T

c

k

Period =2in time domain

w=2.1and w=0.1are the same

cos(2.1n) cos(0.1n)

trigonometric function property

high frequency is changed into low frequency in time domain

w=1.1 and w=0.9are the same

cos(1.1n) cos(0.9n)

trigonometric function property

2.ideal reconstruction

ideal reconstruction in frequency domain

Figure 4.4

s / 2

EXAMPLE Figure 4.5 Take sinusoidal signal for example to

understand aliasing from frequency domain

s 0

EXAMPLE

Sampling frequency:8Hz

Reconstruct frequency:

f ' 8 5 3Hz

Figure 4.10(a) mathematic model for ideal D/C

X r ( j) X s ( j) H r ( j) X c ( j)

T | | c

H r ( j)

0 | | c

X r ( j ) X s ( j ) H r ( j )

1 1 C

jt

hr (t ) IFT {H r ( j)} H r ( j ) e d Te jt d

2 2 C

sin( c t ) sin( t / T )

t / T t / T

sin( t / T )

xr (t ) xs (t ) hr (t ) [ x[n] (t nT )]

n t / T

sin( t / T )

sin[ (t nT ) / T ]

x[n][ (t nT ) ] x[n] ideal reconstruction in

n t / T n (t nT ) / T time domain

sin( x)

sin c( x)

x

sin c( x)

EXAMPLE

Figure 4.9

understand aliasing from time-

domain interpolation

EXAMPLE

3.Nyquist sampling theorems

s N N

s N N

Nyquist sampling theorems:

X c ( j) 0, | | N

then xc (t ) is uniquely det er min ed by its samples

x[n] xc (nT ), n 0,1,2,

if s N N , that is

2 1

( s ) 2 N , or ( f s ) 2 f N

T T

s / 2 : nyquist frequency

2 N : nyquist rate

s 2 N : oversampling

s 2 N : undersampling

Haa ( j)

c s / 2

examples of sampling theorem1

The highest frequency of analog signal ,which wav file with sampling rate

16kHz can show is

8kHz

examples of sampling theorem2

filejudge the sound we can feel

frequency range B

A20~44.1kHz B20~20kHz

C20~4kHz D20~8kHz

EXAMPLE Matlab codes to realize

interpolation

f s 10 Hz(T 0.1s )

draw x[n] xa (nT ) cos(10nT ) cos(n)

draw reconstruction signal :

sin[ (t nT ) / T ]

y (t )

n

x[n]

(t nT ) / T

T=0.1;

n=0:10; x=cos(10*pi*n*T); stem(n,x);

dt=0.001; t=ones(11,1)* [0:dt:1]; n=n'*ones(1,1/dt+1);

y=x*sinc((t-n*T)/T); hold on; plot(t/T,y,'r')

Supplement: band-pass sampling theorem

f s 2( f H f L )(1 M / N )

fH

N int

fH fL

fH

M N

fH fL

f H 5B

4 B f H 5B

4fs

Nf s 2 f H

fH / B (M N )

f s 2 f H / N 2B 2B 2 B(1 M / N )

N N

4.1 summary

x[n] xc (t ) |t nT xc (nT )

2.changes in frequency domain caused by sampling

X ( j( k ))

1

X s ( j) c s

T

k

X ( j( k 2 ) / T )

1

X ( e j ) c

T

k

3. understand reconstruction in frequency domain

4. understand reconstruction in time domain

5. sampling theorem

s N N

s N N

Requirements and difficulties

frequency spectrum chart of sampling and reconstruction

comprehension and application of sampling theorem

4.2 discrete-time processing of

continuous-time signals

Figure 4.11

H (e jT ) | | / T

H eff ( j)

0 | | / T

conditionsLTIno aliasing or aliasing occurred outside the pass band of filters

EXAMPLE

Figure 4.12

EXAMPLE

the pass band of digital

filters satisfies the

equivalent relation of

frequency response

mentioned before.

Figure 4.13

4.3 continuous-time processing of discrete-time signal

Figure 4.16

H (e j ) Hc ( j / T ) for | |

Figure 4.12

EXAMPLE Ideal delay systemnoninteger delay

H (e j ) e j

4.4 digital processing of analog signals

Figure 4.41

Sampling and holding

Figure 4.46(b)

uniform

quantization and

coding

SNR 6 B 1.25(dB)

Figure 4.51

quantization

error of 3BIT

quantization

error of 8BIT

nonuniform quantization

0

vector quantization

x index0

4

4

3 3 3

3 3

2

2 1 1

codeword 1 codeword d(x,c0)=5

c0

c1 d(x,c1)=11

x 4 4 d(x,c2)=8

3 3

2 2

1

d(x,c3)=8

codeword 1 codeword

c2 c3

vector quantization

Example: image coding

Initial image block4 gray-levelsdimentions k=44=16

x 0 1 2 3

d(x,y0)=25

Code book C y0, y1 , y2, y3

d(x,y1)=5

d(x,y2)=25

d(x,y3)=46

y0 y1 y2 y3

codeword y1 is the most adjacent to xso it is coded by

the index 01.

reconstruction

h(t ) RN (t )

Figure 4.5

record the digital sound

Influence caused by sampling rate

and quantizing bits

Different tones require

different sampling rates.

4.1~4.4 summary

frequency domain of sampling and reconstruction.

sampling theorem educed from aliasing in

frequency domain

2. analog signal processing in digital system or digital

signal in analog system , to explain some digital

systems their frequency responses are linear in

dominant period

3. steps in A/D conversion

Requirements and difficulties

sampling processing in time and frequency domain

frequency spectrum chart;

comprehension and application of sampling theorem;

frequency response in discrete-time processing system of

continuous-time signals;

4.5 changing the sampling rate using

discrete-time processing

(downsampling, decimation)

4.5.2 increasing the sampling rate by an integer factor

(upsampling, interpolation)

4.5.3 changing the sampling rate by a noninteger fact

4.5.4 application of multirate signal processing

4.5.1 sampling rate reduction by an integer factor

(downsampling, decimation)

xd [n] x[nM ]

a sampling rate compressor :

Figure 4.20

time-domain of downsampling decrease the datareduce the sampling rate

M=2fs=fs/M,T=MT

X d e M

M 1

1

j ( 2i ) / M

X

i 0

e j ( 2i ) / M

EXAMPLE M=2

j 1

X d (e ) X e j / 2 X e j ( 2 ) / 2)

2

X (e j )

2 0 2

X d ( e j )

1/ 2

2 0 2

Figure 4.21(c)(d)

EXAMPLE M=3

j 1

X d (e ) X e j / 3 X e j ( 2 ) / 3) X e j ( 4 ) / 3)

3

X (e j )

2 0 j 2

X d (e )

2 0 2

EXAMPLE M=3aliasing

Figure 4.22(b)(c)

M times wider1/M times higher

Condition to avoid aliasing N / M

Figure 4.23

4.5.2 increasing the sampling rate by an integer factor

(upsampling, interpolation)

x[n / L] n 0, L,2 L

xe [n]

0 other

or , xe [n] x[k ] [n kL]

k

a sampling rate exp ander :

x e [n]

x[n] L

time-domain of upsampling increase the dataraise the sampling rate

L=2fs=Lfs,T=T/L

X e (e j ) X (e jL )

EXAMPLE

Figure 4.25

L=2

/T' N

L L T

frequency domain of

reverse mirror-image filter

period. Period=2also period =2 /L

total upsampling systemtotal system

Figure 4.24

sin n / L

hi [n] IFT [ H e j ] L

n

xi [n] xe [n] hi [n] ( x[k ] [n kL]) hi [n]

k

x[k ]( [n kL] h [n])

k

i

(n kL)

sin

x[k ]hi [n kL] x[k ]

L

k k

(n kL)

L

EXAMPLE

time-domain process of

mirror-image filter

Use linear interpolation actually

Figure 4.27

4.5.3 changing the sampling rate by a noninteger factor

L

fs ' fs

M

Figure 4.28

EXAMPLE change 400 Hz' s signal to 300 Hz

L 3, M 4

X (e j )

2 0 2

X e (e j ), X i (e j )

2 0 / 4 2

Y ( e j )

2 0 2

1.Combine antialiasing and reverse mirror-image filter

2.Lossless information for upsampling

4.5.4 application of multirate signal processing

rate with low-powered analog antialiasing filter , oversampling and high-powered

digital antialiasing filter, decimation. Transfer the difficulty of the realization of high-

powered analog filter to the design of high-powered digital filter.

Figure 4.43

FIGURE 4.44

2.reconstruction systemreplace high-powered analog reconstructing filter with

interpolation, high-powered digital reverse mirror-image filter and low-powered

analog analog reconstructing filter.

xe[n] xi[n]

x[n] L x(t)

h1(t)

3. filter bank

analysis and synthesis of

sub band

y 0 [ n]

x[n] h0 [n] M M h0 [n] x[n]

y1 [ n]

h1 [ n ] M M h1 [ n ]

..........

..........

..........

..........

.

y N [n]

hN [n] M M hN [n]

In MP3, M=32sub-band analysis filter bank is 32 equi-band filters with center

frequency uniformly distributed from 0 to

MP3 coders use different quantization to realize compression for signals yi[n] in

different sub-bands.

examplecompression for M=2

X (e j )

0 2

Y0 (e j ) Y1 (e j )

' '

0 2

0 2

Y0 (e j ) Y1 (e j )

0 2

0 2

Y0 (e j ) Y1 (e j )

'' ''

0 2 0 2

Y1 (e j )

'

Y0 (e j )

'

0 2 0 2

16bit f s

bit rate before compressio n

16bit fs / 2 8bit fs / 2 12bit fs

bit rate after compressio n

4.pitch scaledecimation or interpolation sampling rate of reconstruction is

not changed.

decimation an d interpolation to

realize pitch scale

4.5 summary

4.5.2 increasing the sampling rate by an integer factor

4.5.3 changing the sampling rate by a noninteger fact

4.5.4 application of multirate signal processing

requirement

frequency spectrum chart of interpolation and decimation

exercises

4.15 (b)(c)

4.24(a)(b)

4.26 only for h= /4

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