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    EE401B/EE3DTR/EE4DTRDigital Transmission

    Pulse Modulation

    Dr John A.R. Williams

    [email protected]

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 1

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    Overview

    Pulse Modulation Schemes

    Pulse Amplitude and Position Modulation

    Time Division Multiplexing

    Pulse Code Modulation

    Quantisation and Companding

    Differential and Delta Pulse Code Modulation

    Adaptive Modulation Techniques

    Subband Coding

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 2

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    Pulse Modulation Schemes

    Pulse PositionModulation(PPM) t

    Pulse FrequencyModulation

    (PFM) t

    t

    tPulse Carrier

    Message Signal

    t

    Pulse AmplitudeModulation

    (PAM)

    t

    Pulse WidthModulation

    (PWM)

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 3

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    Natural Sampling

    0

    t

    (t)

    W W0

    |F( )|

    W = 2B W = 2B0

    t

    (t)

    W W0

    |F( )|

    W = 2B W = 2B

    W W0

    |F( )|

    W = 2B W = 2B

    p(t)

    00

    t

    T 2T 3T2T T

    1

    T Pn

    2/T/T

    0 2

    0

    02

    0

    p(t)

    00

    t

    T 2T 3T2T T

    1

    T PnPn

    2/T/T

    0

    0 2

    02

    0

    0

    02

    02

    0

    2/T

    0 2 0 02 0

    Fs()

    0

    t

    (t)s

    0 T 2T 3T2T T

    T

    2/T

    0 2 0 02 0

    2/T

    0

    0 2 02 0 0 02 02 0

    Fs()Fs()Fs()

    0

    t

    (t)s(t)s

    0 T 2T 3T2T T

    T

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 4

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    Nyquist Sampling Theorem

    1

    T0 W

    T=1/2W

    W 1

    T

    1

    T0 W

    T=1/2W

    W 1

    T

    1

    T

    1

    T

    1

    TW+

    1

    TWW0 W +

    1

    TW

    T1/2W

    W +1

    TW

    1

    TW W

    1

    TW

    Undersampled

    A band-limited signal of finite energy, which has no fre-

    quency components higher than WHz is completely

    described by sampling values of the signal at instantsof time separated by1/(2W)seconds

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 5

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    Pulse Amplitude Modulation - Flat Topped Sampling

    0

    Fs( )

    T

    0

    t

    (t)s

    0

    Fs( )Fs( )

    TT

    0

    t

    (t)s (t)s

    q(t)

    0

    t

    1

    0

    Q()

    -2/ 2/

    q(t)

    0

    t

    q(t)

    0

    t

    1

    0

    Q()

    -2/-2/ 2/

    t

    0

    T(t)s

    q(t)*

    2/T

    0 2 0 02 0 0

    Fs ( ) Q( )

    t

    0

    T(t)s

    q(t)*(t)s

    (t)s

    q(t)* q(t)*

    2/T

    0 0 2 02 0 0 02 02 0 0

    Fs ( ) Q( )Fs ( )Fs ( ) Q( )

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 6

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    PAM Demodulation

    Sample and Hold Circuit

    C R

    T

    Rs

    R sC > TC R

    T

    Rs

    R sC > TC R

    T

    T

    RsRs

    R sC T

    t

    T

    input

    sample-and-holdoutput

    low-pass filteredoutput

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 7

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    Time Division Multiplexing (TDM)

    single communication

    channel carrying N

    signals:

    signal 1

    signal 2

    signal N

    synchronised

    commutatorsignal 1

    signal 2

    signal N

    low-pass filters to

    band-limit signals

    low-pass filters to

    recover signals

    Time Division Multiplexing Of 2 PAM Signals

    t

    TT

    sTT

    sT

    s

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 8

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    TDM Example

    A four channel time division multiplexed PAM system has input signals band-

    limited as follows:

    channel 1: 016 kHz

    channel 2: 020 kHzchannel 3: 025 kHz

    channel 4: 2530 kHz

    If the four channels are sampled at equal time intervals using very short pulses at

    the minimum frequency possible, and the TDM signal is low-pass filtered prior totransmission determine

    1. The minimum clock frequency and the commutator recycling frequency

    2. the minimum cut-off frequency of the LPF consistent with recovery of the

    signals after transmission

    3. What would be the minimum bandwidth required if the four channels were

    frequency division multiplexed using

    (a) DSB-AM

    (b) SSB

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 9

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    TDM Example

    1. Channel 4 has a maximum frequency = 30 kHz and must,

    therefore, be sampled at 60 kHz. This is the maximum frequency

    in the set of channels. Hence for the four channels sampled at

    equal time intervals, the clock rate=460=240kHz and thecommutator must cycle at 60 kHz

    2. For the TDM signal we must have a filter cut-off at half the

    maximum frequencyBc=1/(2Ts) =120kHz, andTs=4.17s

    3. (a) DSB-AM requires2 (16 + 20 + 25 + 5) =132kHz

    (b) SSB requires132/2=66kHz

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 10

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    PPM and PWM Generation

    inputsignal

    Sample

    and hold

    Clock

    Sawtooth

    Generator

    reference level

    PWMComparator

    reference level

    PWMComparator

    PPM

    Pulse

    Generator

    sample-and-holdoutput

    sample-and-holdoutput

    inputsignalinputsignal

    sawtoothsawtooth

    PWMoutputPWMoutput

    PPMoutputPPM

    output

    sumsum

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 11

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    Pulse Code Modulation

    0.0

    0.2

    0.4

    0.6

    0.8

    1.0

    Amplitude

    123

    45678

    9101112131415

    0

    Quantisationlevel

    Binarycode

    1111111011011100101110101001

    10000111011001010100

    0011001000010000

    0111 1101 1110 1010 1001 0101 0011 0101 0111

    Samplinginstants

    sampled signal

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 12

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    PCM Transmission

    PCM Transmitter

    analogueto

    digital converter

    PCM

    signal

    input(analogue)

    signalSampler EncoderQuantiser

    Parallelto

    serial

    converter

    NRZ coded PCM signal with 7-bit samples

    1011001 1110011 0101010

    framing bit

    PCM Receiver

    Output MessageSignal

    DecoderRegenerator Reconstruction

    Filter

    PCM signal+noise

    +distortion

    Output MessageSignal

    DecoderRegenerator Reconstruction

    Filter

    PCM signal+noise

    +distortion

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 13

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    PCM Advantages

    Digital transmission providesrobustnessagainst noise and

    interference

    PCM signals can beregeneratedat intermediate repeaters. Modulating and demodulating circuits areentirely digitaloffering

    compatibility with VLSI: high reliability and low cost

    Signals can be stored in memory;digital signal processingoperationssuch as time scaling can be easily performed

    Encryptionusing special codes can allow secure communication

    Source codingmay be used to avoid unnecessary repetitions offrequent message

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 14

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    Quantisation

    (t)=f(t)-f (t)

    f(t)

    q

    t

    /2

    /2

    f (t)q

    0

    0

    Error

    Magni

    tude

    0

    1

    2

    3

    4

    M-3

    M-2

    M-1

    M

    M

    2

    +1M2

    V+

    2

    V

    2

    True SignalLevel

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 15

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    Quantisation Noise

    Assume all values in the range /2< < /2are equally probable. p() =1/.Then

    2

    =

    1

    Z /2

    /2

    2

    d=

    1

    3

    3/2/2 =

    2

    12

    If a signal occupies thefull quantiser range. Peak signal amplitude isM/2giving

    (SNRQ)peak=(M/2)2

    2/12 =3M2 =322n

    (SNRQ)peak=10{log10 3 + 2n log10 2} =4.8 + 6n[dB]

    If we have asinusoidal signal, mean signal power is(M/2)2/2and

    (SNRQ)mean=10{log10 1.5 + 2n log10 2} =1.8 + 6n[dB]

    TheSNRQincreases by 6 dB for each additional bit used in quantisation

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 16

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    Quantisation Example

    What is the quantisation noise for the case where the quantiser always rounds down to

    the next lowest level (i.e. the maximum error is).

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 17

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    Quantisation Example

    What is the quantisation noise for the case where the quantiser always rounds down to

    the next lowest level (i.e. the maximum error is).If we round down the quantisation error is equiprobable in the range 0

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    Companding

    A-law compression characteristic

    1/A1-1 0

    -1/A

    Vout

    Vin1/A

    1-1 0

    -1/A

    Vout

    Vin

    Vout= AVin

    1 + logeA , 0 |Vin|

    1

    A (linear region)

    =1 + loge(AVin)

    1 + logeA,

    1

    A |Vin| 1, (logarithmic region)

    A=87.7in Europe

    Vout

    Vin

    Increasing A

    Vout

    Vin

    Increasing A

    0

    1/A

    2dB {

    Vin

    SNRQ

    WithoutCompanding

    0

    1/A

    2dB {

    Vin

    SNRQ

    WithoutCompanding

    0

    1/A

    2dB {

    Vin

    SNRQ

    WithoutCompanding

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 18

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    Segmented Companding

    640 128 256

    512 1024 2048 4096

    32

    48

    64

    80

    96

    112

    128

    0

    Input Level

    128 16

    64 32

    64 16

    256 16512 16

    1024 16

    2048 166 bi ts equiv.

    10 bits equiv.

    12 bits equiv.

    11 bits equiv.

    9 bi ts equiv.

    8 bi ts equiv.

    7 bi ts equiv.

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 19

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    NICAM

    Nearly Instantaneous Companded Audio Multiplex

    BitsRange 0

    MSB LSB

    Range 1Range 2Range 3Range 4

    1 2 3 4 5 6 7 8 91011121314

    Transmit ted bits

    14 bit resolution with 5 ranges represented in 3 bits

    1 range setting per block of 32 samples (1 ms)

    Ranges packed into 7 bits every 3 blocks (3ms)

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 20

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    Differential Pulse Code Modulation

    If we oversample changes in signal amplitude are small and we can use

    fewer bits to quantise.

    SampledInput Signal

    InputSignal

    Range

    DifferentialSignalRange

    t

    t

    Ts

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 21

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    Differential Pulse Code Modulation

    DPCM Encoder

    Sampler

    Predictor

    + Quantiserx(t) xn

    xn

    +

    en en

    {ai}

    en=xn

    p

    i=1

    aixni

    DPCM Decoder

    +

    Predictor

    en

    xn

    xn

    {ai}

    nx= en+

    p

    i=1

    ai xni

    xnxn =

    en+

    p

    i=1 ai xni

    en+

    p

    i=1 aixni

    = qn+p

    i=1

    ai( xnixni)

    i.e. there is an accumulation of quantisation errors at the receiver.

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 22

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    DPCM Processing Gain

    Sampler

    Predictor

    + Quantiserx(t) xn

    xn

    +

    en en

    {ai}

    SNRO=2x

    2

    E

    2E

    2

    Q

    =GPSNRQ

    Minimise2Eusing Yule-Walker equations

    p

    i=1 ai(i j) =(j), j=1,2, . . . ,p

    where (n)is the autocorrelation function of the sampled signal sequence xn, whichmay be estimated from the finite set of samples {xn} by

    (n) = 1

    N

    Nn

    i=1

    xixi+n,n=0,1, . . . ,p

    (See Proakis Page 128)

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 23

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    Practical DPCM

    DPCM Encoder

    Sampler + Quantiser

    Predictor +

    x(t) xn +

    en en

    xnxn

    {ai}

    en=xnp

    i=

    1

    ai xni

    DPCM Decoder

    +

    Predictor

    en

    xn

    xn= xn+ en

    {ai}

    To lowpass

    Filter

    xn= en+p

    i=1

    ai xni

    xnxn =

    en+

    p

    i=1

    ai xni

    en+

    p

    i=1

    ai xni

    = qn

    Error at receiver is only the quantisation error.c 20012008 Dr John A.R. Williams Pulse Modulation p. 24

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    Improved DPCM

    Encoder

    Sampler + Quantiser

    +Linear Filter

    {ai}

    Linear Filter{bi}

    +

    x(t) xn +

    en en

    xnxn

    Decoder

    +

    +Linear Filter

    {ai}Linear Filter

    {bi}

    en xn To lowpass

    Filter

    Improve estimate us-

    ing linearly filtered past

    values of the quantised

    error.

    xn =

    p

    i=1

    ai xni

    +m

    i=1

    bieni

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 25

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    Delta Modulation (DM)

    DM Encoder

    Unit Delayz-1

    QuantiserSamplerx t( ) xn en

    ~xn

    ~$ ~x xn n= 1

    +

    -

    ~en = 1 DM Decoder

    Z-1

    ~en ~ ~$x x en n n= + To lowpassfilter

    Ts

    t

    1 1 0 1 1 0 1 0 1 1 0 1 0 1 0 0 0 0 0 0 0 1 0 1 0 1 0 1 1 1

    Input Signal ( )f t

    StepOverload

    A simple practical approach

    +

    f t( )Comparator

    BinaryOutputSignal

    Clock

    Flip-flop

    (Output v)

    R

    C

    R

    C

    RC Integrator

    +V

    -V

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 26

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    Variable Step Size DM

    DM Distortion

    t

    Slope-

    overloaddistortion

    Granular Noise

    x t( )

    t

    Slope-

    overloaddistortion

    Granular Noise

    x t( )

    With Variable Step Size

    t

    Slope-overloaddistortion

    Granular Noisex t( )

    Adaptive DM Encoder

    QuantiserSampler

    Accumulator

    x t( ) xn en+

    -

    ~en = 1

    ~en1

    z-1

    z-1

    To transmitter

    n

    n1

    Adaptive DM Decoder

    AccumulatorOutput

    LowpassFilter

    ~en

    ~en1

    z-1

    z-1

    n

    n1

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 27

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    Slope Overload Distortion Example

    Consider a sine wave of frequency fmwith amplitudeAmapplied to a

    delta modulator of step size. Show that the slope-overload distortionwill occur if

    Am>

    2fmTs

    whereTsis the sampling period. What is the maximum power that may

    be transmitted without slope overload distortion?

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 28

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    Slope Overload Distortion Example

    Consider a sine wave of frequency fmwith amplitudeAmapplied to a

    delta modulator of step size. Show that the slope-overload distortionwill occur if

    Am>

    2fmTs

    whereTsis the sampling period. What is the maximum power that may

    be transmitted without slope overload distortion?Modulated Wavem(t) =Am cos2fmthas slopedm(t)

    dt = 2fmAm sin2fmt.

    Maximum average slope reproduced by the delta modulator is/Ts.Therefore require2fm>

    Ts

    orAm>

    2fmTs.

    Maximum power is A2m

    2 =

    2

    8f2m

    T2

    s

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 28

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    Adaptive Quantisation

    000

    M(4)

    001

    M(3)

    010

    M(2)

    011

    M(1)

    100

    M(1)

    101M(2)

    110

    M(3)

    111

    M(4)

    0

    -3 -2 -1

    1 2 3Input

    Output Previous

    output

    Multiplier

    7

    2

    12

    52

    32

    32

    52

    12

    72

    EncoderInput

    EncoderOutput

    LevelEstimator

    LevelEstimator

    Quantiser with an adaptive step

    N.S. Jayant, Digital Coding of

    Speech Waveforms: PCM, DPCM,

    and DM Quantizers, Proc IEE, vol62, pp. 611632.

    PCM DPCM

    2 3 4 2 3 4

    M(1) 0.60 0.85 0.80 0.80 0.90 0.90

    M(2) 2.20 1.00 0.80 1.60 0.90 0.90

    M(3) 1.00 0.80 1.25 0.90

    M(4) 1.50 0.80 1.70 0.90

    M(5) 1.20 1.20

    M(6) 1.60 1.60

    M(7) 2.00 2.00

    M(8) 2.40 2.40

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 29

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    Adaptive Prediction

    Calculate predictor coefficients from short term estimate of the autocorrelation

    function ofxn.

    Receiver predictor may compute its own predictor coefficients from signal esti-

    mates provided quantisation error is small.

    Adaptive Prediction withbackward estimation (APB)

    Predictor

    Logic foradaptive

    Prediction

    QuantiserSamplerx t( ) xn en

    ~en

    ~xn~$

    xn

    +

    -

    Adaptive Differential Pulse-Code Modulation (ADCPM) combines adaptive

    quantisation and Prediction.

    A standard for speech encoded transmission at 32 kb/s. Standard PCM requires

    64 kb/s for speech encoded transmission.

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 30

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    Adaptive Subband Coding

    exploit quasi-periodic nature

    of voiced speech permitting

    pitch prediction reducing thelevel of prediction error

    requiring quantisation

    human ear cannot hear noise

    below 15 dB below thesignal level in same band

    can digitise speech at 16 kb/s

    with a quality comparable to

    64 kb/s PCM.

    FilterBank f or

    SubbandAnalysis

    Adaptivebi t

    assignmentcircuit

    Multiplexer

    Speech

    Signal

    To

    channel

    ADPCM Encoders

    FilterBank f orSubbandAnalysis

    De-multiplexer

    Fromchannel

    Output

    ADPCM Decoders

    c 20012008 Dr John A.R. Williams Pulse Modulation p. 31