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A Periodic Signal X (T) With A Period of T Is Expanded As A Fourier Series

The document discusses Fourier series analysis and sampling of periodic signals. 1) A periodic signal can be represented as a Fourier series expansion consisting of a DC term plus cosine and sine terms with coefficients determined by integrating the signal over one period. 2) The Fourier series of a sine wave consists solely of a sine term with the fundamental frequency. 3) Nyquist-Shannon sampling theorem states that a bandlimited analog signal can be perfectly reconstructed from its samples if the sampling frequency is at least twice the maximum frequency of the signal. 4) Aliasing occurs if the sampling frequency is too low and high frequency components are misinterpreted as lower frequencies in the reconstructed signal. Anti-aliasing filters are

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Seungnam Kim
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55 views

A Periodic Signal X (T) With A Period of T Is Expanded As A Fourier Series

The document discusses Fourier series analysis and sampling of periodic signals. 1) A periodic signal can be represented as a Fourier series expansion consisting of a DC term plus cosine and sine terms with coefficients determined by integrating the signal over one period. 2) The Fourier series of a sine wave consists solely of a sine term with the fundamental frequency. 3) Nyquist-Shannon sampling theorem states that a bandlimited analog signal can be perfectly reconstructed from its samples if the sampling frequency is at least twice the maximum frequency of the signal. 4) Aliasing occurs if the sampling frequency is too low and high frequency components are misinterpreted as lower frequencies in the reconstructed signal. Anti-aliasing filters are

Uploaded by

Seungnam Kim
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© Attribution Non-Commercial (BY-NC)
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Download as DOCX, PDF, TXT or read online on Scribd
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A periodic signal x(t) with a period of T is expanded as a fourier series

Ex.1

Given : Period T = (

) sec ,

()

()

( )

()

( )

Hence the Fourier series is () ( ) ( )

..

How about F.S. of sin t : Sol:

T , (sec)

Dc :

() ()

() - ( )-

, ,

Cos term: n =1,2,


() () () (

( )

) (

( )

Since ( ) ( ),

implies

(( 0 ) ((

))

(( ) ) - )( ))-

)) (( , (( )) - 1 ) ) (( )( ))+,

((

Since ( )

),

Sin term:

() () () (

( )

) (

( )

Since (

) (( ))

( (( ((

), implies )) )) - 1 )), (( ) ) (( )( ))+

* *

0 ,

(( (( ,

) ) - ) ) ((

)( , , ,

Since

For n = 2,3,4,., bn= 0. However, For n = 1, since the denominator is also zero, we have to cal,.. ** (l'Hpital's rule, l'Hospital's rule) f(x) , g(x) x = c , ( ) ( )

( ) ( )

** Hence , for n= 1,

* *

(( ((

) ) ) ) * ((

(( ((

)(

))+

) ( ))+

) )+

((

) )+

Therefore for sin(t) , the F.S. is , for all n

Sin(t)

=sin(t) Intuitively it is correct

** re visit (l'Hpital's rule, l'Hospital's rule) ,

The other method, using Tayler series

Since

, Hence

********************

2.SIGNAL SAMPLING,

Figure 2.1

2.1 Sampling Definition: Sampling : - : , -

-In signal processing, sampling is the reduction of a continuous signal to a discrete signal. Example : Video sampling movie. 1 24 , TV 30 . 24 / sampling 1 24 / (24Hz) . rotation speed,.. led, http://www.youtube.com/watch?v=oIX-TsN4SZ4&feature=more_related
One application :

stroboscope : objects are moving continuously camera , flash lamp,.. In electrical signal processing To have digital data, -Why digital data 1) noise robust 2) . s/w radio : AM/FM/TV/ , 3)memory -continuous signal ??? LP (long playing (memory) cassette tape disc,.. flash memory : 1 tera ( ??)

SO except power, all signal be digital. How to get digital signal from analog signal ? Analog Sampling Holding Quantization coding digital 1)sampling

Given ( ), sampling with T , get digital signal , - such as

, -

),

, , ,

, , ,

T : -in general it is fixed not changed .. Sampling frequency ( Hz, or 1/sec) ( rad/sec)

Sampling angular frequency Ex: sampling period T = 1 micro sec T= 125 micro sec

2)Holding

3)Quantization 4)coding (real value x bits binary code)

Sampling theorem: Problem : Given digital signals , how to get original analog nsignal?? :

, -

),

, , ,

, , ,

x(t) , digital discrete analog ??

Nyquist-Shannon sampling theorem: -If the analog signal is band limited, ( F(f) = 0, where f>fmax) -If sampling rate fs is as large as the twice of the max of fmax, (fs >= fmax ) Then the original analog can be recovered by the digital signal. !!analog . !!stroboscope Example

: Sampling rate : 0.01sec fs = 1/0.01= 100(Hz) Original : T = 0.1/4=0.025 sec

f=fmax=1/0.025=40(Hz) sampling . : Sampling rate : 0.01sec fs = 1/0.01= 100(Hz) Original : f=fmax=90(Hz) sampling !!: : Sampling rate : 0.01sec fs = 1/0.01= 100(Hz) Original : f=fmax=10(Hz) sampling !!: If , sample 100Hz 90Hz , 10hz Alias!! **Alias :

: Given band limited con. ( ) , pulse train sampled signal () ( ), () ( ), , then

**Eulers Identity

where And

=-1.

( )

) **

** F.series two-side frequency domain **

()

()

()

A: given band limited signal, B=fmax B,C,D:sampled signal With a proper filter, we get the original signal x(t) from xs(t) !! ( only sampled data signal !! analog version: ,... digital version : (modem), B: Nyquist C: just D : alias **oscilloscope :sampling rate

60Mhz,100Mhz,.10Ghz, undersampling , alias , recover .*** ** under sampling *** ** , maximum band , sampling rate ? ,.. ,.. ::sampling rate ?? -- --memory sampling rate 24Hz..100Hz

2.2.1 practical consideration: Anti-Aliasing filter Problem: given sampling rate, fs, in order for the recovered signal not to be alias, design the filter .. The output of the anti-alias filter is of max band width less than half of fs the filters bandwidth <= 1/2 fs

In the text book, analogue filter designed Butter worth( Sallen-Key low pass filter), -

2.2.2 anti-image filter and equalizer Holding circuit : ( ) is a step function , such as () define h(t) as () () ( ) ,

. The Laplace transform of h(t) is

( )

**Laplace transform : for ( ), ( ) Ex. Ex.1 ( ) ()

( )

Ex.2 Let

( ) is the L. of ( ), fine the L. of

()

()

Sol :

( )

( ()

))

Since

()

Let , then

()

()

Therefore ( )

()

()

()

( () ( )

( ( )

))

HW.1 If ( )

, how about x(t)-x(t-T) ??

HW.2 derive 2.13, 2.14., 2.15,2.17,2.18

**dB,,, 1.Power measurement (Watt,)

Ex.. 0 dBm = 1mW

Ex. SMT-W6100 IP phone : : /== 2.Amplitude !!

Ex..page 34 Mag. of the transfer function -0.46dB ************************** ( ), , ,

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