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I.1 Sampling Theorem and Pulse Amplitude Modulation (PAM) Reference Stremler, Communication Systems, Chapter 3.15, 7.1 I.2 Sampling Theorem Signals bearing information are either in analog form, discrete form or digital form. Sampling theorem determines the necessary conditions which allow us to change an analog signal to a discrete one, or vice versa, without loss of information. Sample and Hold encoder Analog-to-digital converter 10110..

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  • 1I.1

    Sampling Theorem and Pulse Amplitude Modulation(PAM)

    Reference Stremler, Communication Systems, Chapter 3.15, 7.1

    I.2

    Sampling Theorem

    Signals bearing information are either in analog form,discrete form or digital form.

    Sampling theorem determines the necessary conditionswhich allow us to change an analog signal to a discrete one,or vice versa, without loss of information.

    Sampleand Hold

    encoder

    Analog-to-digital converter

    10110..

  • 2I.3

    Sampling Theorem:A real-valued band-limited signal having no spectralcomponents above a frequency of B Hz is determineduniquely by its values at uniform interval spaced no greaterthan 1/(2B) second apart.

    Bt

    f(t) F() fs(t)

    Freq (Hz) tT

    BT

    21

    (Hz)bandwidth signal:(second) period sampling:

    BT

    i.e. taking more than or equal to 2B samples in each second

    I.4

    Example:To convert a 10kHz sinusoidal signal to digital form, theminimum sampling frequency is 20kHz.

    To convert a voice signal (0-3.3kHz) to digital form, theminimum sampling frequency is 6.6kHz.In practice, sampling frequency = 8kHz.

    To convert an audio signal (0-20kHz) to digital form, theminimum sampling frequency is 40kHz.In practice, sampling frequency for encoding music into CDis 44.1kHz.

  • 3I.5

    This is a sufficient condition such that an analog signalcan be reconstructed completely from a set of uniformlyspaced discrete samples in time.

    ProofConsider a band-limited signal f(t) having no spectralcomponents above B Hz.

    2Bt

    f(t) F()-2B

    I.6

    The signal is sampled using the periodic gate function pT(t).As pT(t) is a periodic signal, it can be represented by aFourier series.

    )(tpT

    T T2T t

    nP

    )/(

    /2)(

    TnSaT

    P

    TePtp

    n

    on

    tjnnT

    o

    =

    == =

    oo2

    xxxSa sin)( =

  • 4I.7

    The sampled signal is

    Taking the Fourier transform, we have

    )(tfs

    =

    ==

    n

    tjnn

    Ts

    oePtf

    tptftf

    )(

    )()()(

    { })propertyn translatiofrequency ()(

    )Linearity()(

    )()(

    on

    n

    tjn

    nn

    n

    tjnns

    nFP

    etfFP

    ePtfFF

    o

    o

    =

    =

    =

    =

    =

    =

    )(tfs

    t

    )(sF

    oo2

    )(FPo )(1 oFP

    Not a function of

    I.8

    Therefore, the spectral density of the sampled signal is,within a constant factor, exactly the same as that of . Inaddition, it repeats itself periodically. The spectral density ofthe original signal can be retrieved by using a LPF on .

    However, if the sampling period T >1/2B, the replicas ofwill overlap and we cannot retrieve from .

    )(tfs)(tf

    )(sF

    )(F )(sF)(F

  • 5I.9

    )(sF

    BTo 4/2 >=

    )(FPo )(sF

    BTo 4/2 ==

    )(FPo

    )(sF

    To /2 =

    )(sF

    BTo 4/2

    I.10

    The maximum time interval T of sampling (=1/2B) is calledthe Nyquist interval; its reciprocal (2B) is called the Nyquistsampling frequency.

    In practice, oversampling (T < 1/2B) is used. we cannot build ideal lowpass filter. If the filter

    characteristics has a finite slope at the band edges,frequency components from the spectral replicas may betransmitted through the filter.

  • 6I.11

    A time-limited signal is never strictly band-limited. Whensuch a signal is sampled, there will be some unavoidableoverlap of spectral components. In reconstruction of thesignal, frequency components originally located aboveone-half the sampling frequency will appear below thispoint. This is known as aliasing.

    )(sF

    To /2 =

    I.12

    Pulse Amplitude Modulation (PAM)

    In pulse amplitude modulation (PAM) the amplitude of atrain of constant-width pulses is varied in proportional to thesample values of the modulating signal.

    f(t))(tfPAM

    t t

  • 7I.13

    Generating a PAM signal could be divided into twoprocesses: sampling and holding

    Sampling: Consider a lowpass signal that is band-limited to and multiplied by a periodic train of verynarrow rectangular pulses . The sampling interval T istaken as the Nyquist interval seconds.

    )(tfmf

    )(tpTmf2/1

    f(t) F()

    mm)(tpT

    t

    t

    )(TP

    T T/2

    I.14

    The sampled signal is

    Taking the Fourier transform, we have

    TPTePtf

    tptftf

    non

    tjnn

    Ts

    o /1 and /2 where)(

    )()()(

    ====

    =

    { }

    { }

    =

    =

    =

    =

    =

    =

    =

    no

    tjn

    nn

    n

    tjnn

    Ts

    nFT

    etfFP

    ePtfF

    tptfFF

    o

    o

    )(1

    )(

    )(

    )()()(

  • 8I.15

    fs(t)

    t

    )(sF

    T/2

    Holding (lengthening): achieved by applying the sampledsignal to a time-invariant filter with unit impulse response

    )(tf )(tq )(tfPAM)(tfs

    t)(tq

    I.16

    )(Q

    /2

    )2/()( SaQ =

    [ ]

    )()()(

    )()()(

    )()()()()()(

    tqnTtnTf

    tqnTttf

    tqtptftqtftf

    n

    n

    T

    sPAM

    =

    ==

    =

    =

    =

    [ ]

    =

    =

    =

    =

    n

    n

    nTtqnTf

    tqnTtnTf

    )()(

    )()()(

    )(tfPAM

    t

  • 9I.17

    The spectral density of the PAM signal is{ }

    =

    ==

    =

    no

    s

    sPAM

    QnFT

    QFtqtfFF

    )()(1)()(

    )()()(

    )(PAMF /2

    ( ) )(1 QFT

    I.18

    The spectrum obtained here is not the same as thatobtained in I.7. In I.7 the spectrum consists of and its replicas at

    multiples of the sampling frequency with only a gainvariation of each spectral replica (i.e. ).

    The present spectrum describes a point-to-pointmultiplication in frequency so that the spectral densityhas lost its original shape (i.e. ). Thisdistortion is dependent on the pulse shape; at lowfrequencies it is not severe if the pulse width is verynarrow.

    )(F

    )(F

    )(0 FP

    )()( FQ