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Applicat
   ion of
    fourier
    series
      in

SAMPLING

Presented by:
                GIRISH DHARESHWAR
WHAT IS SAMPLING ?
• It is the process of taking the
   samples of the signal at intervals

                            Aliasing
                            cannot distinguish between
                             higher and lower frequencies

                            Sampling theorem:
                             to avoid aliasing, sampling rate
                            must be at least twice the
                            maximum frequency component
                            (`bandwidth’) of the signal
• Sampling theorem says
  there is enough
  information to reconstruct
  the signal, which means
  sampled signal looks like
  original one
Why ??????????
• Most signals are analog in
  nature, and have to be sampled
 loss of information
• Eg :Touch-Tone system of
  telephone dialling, when button
  is pushed two sinusoid signals
  are generated (tones) and
  transmitted, a digital system      speech signal
  determines the frequences and
  uniquely identifies the button –
  digital
Where ???IN COMMUNICATION
A AO
 NL G                  D ITA
                        IG L                 D ITA
                                              IG L

        SML G
        A P IN                     DP
                                    S
S NL
 IG A                  S NL
                        IG A                  S NL
                                               IG A


• Convert analog signals into the digital information-
sampling & involves analog-to-digital conversion
D ITA
 IG L          D ITA
                IG L                            A AO
                                                 NL G

        DP
         S     S NL
                IG A
                           R C N TR C N
                            E O S U TIO
S NL
 IG A                                           S NL
                                                 IG A



 convert the digital information, after being processed
   back to an analog signal
• involves digital-to-analog conversion & reconstruction
 e.g. text-to-speech signal (characters are used to
 generate artificial sound)
AA G
N LO                              D ITA
                                   IG L                   AA G
                                                          N LO
                   D ITA
                    IG L
          S MP G
           A LIN    S NL
                     IG A
                            DP
                             S     S NL
                                    IG A
                                           R C N TR C N
                                            E O S U TIO
                                                          S NL
                                                           IG A
S NL
 IG A




       perform both A/D and D/A conversions

 e.g. digital recording and playback of music (signal is
  sensed by microphones, amplified, converted to digital,
  processed, and converted back to analog to be played
Sampling rate :

8

                                                                   5*sin (2 4t)
6



4
                                                                   Amplitude = 5

2                                                                  Frequency = 4 Hz

0



-2



-4



-6
                                                                         We take an
-8
     0   0.1   0.2   0.3   0.4   0.5   0.6   0.7   0.8   0.9   1
                                                                         ideal sine wave
                                                                         to discuss
                                                                         effects of
                       A sine wave
                                                                         sampling
A sine wave signal and correct sampling
     8

                                                                               5*sin(2 4t)
     6

                                                                               Amplitude = 5
     4



     2
                                                                               Frequency = 4 Hz

     0                                                                         Sampling rate = 256
                                                                               samples/second
    -2

                                                                               Sampling duration =
    -4
                                                                               1 second
    -6


                                                                              We do sampling of 4Hz
    -8
         0   0.1   0.2   0.3   0.4     0.5     0.6   0.7   0.8     0.9   1
                                                                              with 256 Hz so sampling
                                     seconds
                                                                              is much higher rate than
                                                                              the base frequency, good

                                                                 Thus after sampling we can reconstruct
                                                                 the original signal
Here sampling rate is 8.5 Hz
and the frequency is 8 Hz
                                             An undersampled signal
                                                                        Sampling rate

                                                                                                  Red dots
                        2
                                                   sin(2 8t), SR = 8.5 Hz                         represent the
                                                                                                  sampled data
                      1.5



                        1



                      0.5



                        0


                                                                                                  Undersampling
                      -0.5
                                                                                                  can be confusing
                       -1                                                                         Here it suggests
                                                                                                  a different
                      -1.5
                                                                                                  frequency of
                       -2
                             0   0.2   0.4   0.6      0.8    1    1.2       1.4   1.6   1.8   2
                                                                                                  sampled signal


       Loss of information
The Discrete Time Fourier Transform
(DTFT) and its Inverse :

• The Fourier transform is an equation to
  calculate the frequency, amplitude and phase
  of each sampled signal needed to make up
  any given signal f(t):

   F (    )          f (t ) e x p (   i   t ) dt



               1
 f (t )               F (    ) ex p (i    t) d
              2
(t)
function Properties

                                                                         t
  (t ) d t   1


  (t    a ) f (t ) d t               (t   a ) f (a ) dt   f (a )


ex p ( i t ) d t         2       (


ex p [ i (          ') t ] d t        2      (        '
The Fourier Transform of                   (t) is 1.
                   ( t ) exp( i t ) dt     exp( i [0])     1


                          (t)



                                t


And the Fourier Transform of 1 is   ( ):       1 exp( i t ) dt   2   (


                                                         ( )


                                t
The Fourier transform of exp(i 0 t)

   F        exp( i   0
                         t)          exp( i    0
                                                   t ) exp( i    t ) dt


                          exp( i [       0
                                             ] t ) dt       2    (        0
                                                                              )



                exp(i 0t)
                                                        F   {exp(i 0t)}
       Im                            t

       Re                            t




The function exp(i 0t) is the essential component of Fourier analysis. It is
a pure frequency.
The Fourier transform of cos(                                             t)
  F   cos(       0
                     t)              cos(         0
                                                      t ) exp( i          t ) dt


             1
                            exp( i   0
                                             t)    exp( i            0
                                                                         t ) exp( i     t ) dt
             2

      1                                                          1
                 exp( i [                0
                                             ] t ) dt                      exp( i [          0
                                                                                                  ] t ) dt
      2                                                          2

             (              0
                                )                  (         0
                                                                 )

                          cos( 0t)                                                 F {cos(       t )}
                                                                                             0


                                      t
The Modulation Theorem: The Fourier
Transform of E(t) cos( 0 t)

F   E ( t ) cos(     0t )              E ( t ) cos(            0t )   exp( i          t ) dt

                      1
                                 E ( t ) exp( i         0t )          exp( i         0t )   exp( i     t ) dt
                      2
         1                                                             1
                E ( t ) exp( i [          0 ] t ) dt                               E ( t ) exp( i [        0 ]t)   dt
         2                                                                 2

                                          1                                         1 
    F        E ( t ) cos(      0t )         E(                        0)               E(             0)
                                          2                                          2

                                            F           E ( t ) cos(       0t )

        If E(t) = (t), then:
                                                -   0                          0
The Fourier transform and its inverse are symmetrical:
f(t) -> F( ) and F(t) -> f( ) (almost).
If f(t) Fourier transforms to F( ), then F(t) Fourier transforms to:

                             F (t ) ex p (        i    t ) dt

Rearranging:                                                                  2   f(   )
                                 1
                         2                F ( t ) e x p ( i[        ] t) dt
                                 2

Relabeling the integration variable from t to ’, we can see that we have an
inverse Fourier transform:
                             1
                     2               F(        ) exp( i[       ]   )d
                             2

                                           2     f(        )
This is why it is often said that f and F are a “Fourier Transform Pair.”
Summary
•     Fourier analysis for periodic functions focuses on the
    study of Fourier series
•    The Fourier Transform (FT) is a way of transforming a
    continuous signal into the frequency domain
•    The Discrete Time Fourier Transform (DTFT) is a
    Fourier Transform of a sampled signal
•   The Discrete Fourier Transform (DFT) is a discrete
    numerical equivalent using sums instead of integrals
    that can be computed on a digital computer
•    As one of the applications DFT and then Inverse DFT
    (IDFT) can be used to compute standard convolution
    product and thus to perform linear filtering
Application of fourier series

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Application of fourier series

  • 1. Applicat ion of fourier series in SAMPLING Presented by: GIRISH DHARESHWAR
  • 2. WHAT IS SAMPLING ? • It is the process of taking the samples of the signal at intervals Aliasing cannot distinguish between higher and lower frequencies Sampling theorem:  to avoid aliasing, sampling rate must be at least twice the maximum frequency component (`bandwidth’) of the signal
  • 3. • Sampling theorem says there is enough information to reconstruct the signal, which means sampled signal looks like original one
  • 4. Why ?????????? • Most signals are analog in nature, and have to be sampled  loss of information • Eg :Touch-Tone system of telephone dialling, when button is pushed two sinusoid signals are generated (tones) and transmitted, a digital system speech signal determines the frequences and uniquely identifies the button – digital
  • 5. Where ???IN COMMUNICATION A AO NL G D ITA IG L D ITA IG L SML G A P IN DP S S NL IG A S NL IG A S NL IG A • Convert analog signals into the digital information- sampling & involves analog-to-digital conversion D ITA IG L D ITA IG L A AO NL G DP S S NL IG A R C N TR C N E O S U TIO S NL IG A S NL IG A convert the digital information, after being processed back to an analog signal • involves digital-to-analog conversion & reconstruction e.g. text-to-speech signal (characters are used to generate artificial sound)
  • 6. AA G N LO D ITA IG L AA G N LO D ITA IG L S MP G A LIN S NL IG A DP S S NL IG A R C N TR C N E O S U TIO S NL IG A S NL IG A  perform both A/D and D/A conversions  e.g. digital recording and playback of music (signal is sensed by microphones, amplified, converted to digital, processed, and converted back to analog to be played
  • 7. Sampling rate : 8 5*sin (2 4t) 6 4 Amplitude = 5 2 Frequency = 4 Hz 0 -2 -4 -6 We take an -8 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 ideal sine wave to discuss effects of A sine wave sampling
  • 8. A sine wave signal and correct sampling 8 5*sin(2 4t) 6 Amplitude = 5 4 2 Frequency = 4 Hz 0 Sampling rate = 256 samples/second -2 Sampling duration = -4 1 second -6 We do sampling of 4Hz -8 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 with 256 Hz so sampling seconds is much higher rate than the base frequency, good Thus after sampling we can reconstruct the original signal
  • 9. Here sampling rate is 8.5 Hz and the frequency is 8 Hz An undersampled signal Sampling rate Red dots 2 sin(2 8t), SR = 8.5 Hz represent the sampled data 1.5 1 0.5 0 Undersampling -0.5 can be confusing -1 Here it suggests a different -1.5 frequency of -2 0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 sampled signal  Loss of information
  • 10. The Discrete Time Fourier Transform (DTFT) and its Inverse : • The Fourier transform is an equation to calculate the frequency, amplitude and phase of each sampled signal needed to make up any given signal f(t): F ( ) f (t ) e x p ( i t ) dt 1 f (t ) F ( ) ex p (i t) d 2
  • 11. (t) function Properties t (t ) d t 1 (t a ) f (t ) d t (t a ) f (a ) dt f (a ) ex p ( i t ) d t 2 ( ex p [ i ( ') t ] d t 2 ( '
  • 12. The Fourier Transform of (t) is 1. ( t ) exp( i t ) dt exp( i [0]) 1 (t) t And the Fourier Transform of 1 is ( ): 1 exp( i t ) dt 2 ( ( ) t
  • 13. The Fourier transform of exp(i 0 t) F exp( i 0 t) exp( i 0 t ) exp( i t ) dt exp( i [ 0 ] t ) dt 2 ( 0 ) exp(i 0t) F {exp(i 0t)} Im t Re t The function exp(i 0t) is the essential component of Fourier analysis. It is a pure frequency.
  • 14. The Fourier transform of cos( t) F cos( 0 t) cos( 0 t ) exp( i t ) dt 1 exp( i 0 t) exp( i 0 t ) exp( i t ) dt 2 1 1 exp( i [ 0 ] t ) dt exp( i [ 0 ] t ) dt 2 2 ( 0 ) ( 0 ) cos( 0t) F {cos( t )} 0 t
  • 15. The Modulation Theorem: The Fourier Transform of E(t) cos( 0 t) F E ( t ) cos( 0t ) E ( t ) cos( 0t ) exp( i t ) dt 1 E ( t ) exp( i 0t ) exp( i 0t ) exp( i t ) dt 2 1 1 E ( t ) exp( i [ 0 ] t ) dt E ( t ) exp( i [ 0 ]t) dt 2 2 1  1  F E ( t ) cos( 0t ) E( 0) E( 0) 2 2 F E ( t ) cos( 0t ) If E(t) = (t), then: - 0 0
  • 16. The Fourier transform and its inverse are symmetrical: f(t) -> F( ) and F(t) -> f( ) (almost). If f(t) Fourier transforms to F( ), then F(t) Fourier transforms to: F (t ) ex p ( i t ) dt Rearranging: 2 f( ) 1 2 F ( t ) e x p ( i[ ] t) dt 2 Relabeling the integration variable from t to ’, we can see that we have an inverse Fourier transform: 1 2 F( ) exp( i[ ] )d 2 2 f( ) This is why it is often said that f and F are a “Fourier Transform Pair.”
  • 17. Summary • Fourier analysis for periodic functions focuses on the study of Fourier series • The Fourier Transform (FT) is a way of transforming a continuous signal into the frequency domain • The Discrete Time Fourier Transform (DTFT) is a Fourier Transform of a sampled signal • The Discrete Fourier Transform (DFT) is a discrete numerical equivalent using sums instead of integrals that can be computed on a digital computer • As one of the applications DFT and then Inverse DFT (IDFT) can be used to compute standard convolution product and thus to perform linear filtering