Chapter 11 Digital Data Transmission by Baseband Pulse Amplitude Modulation (PAM) Contents

Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 Pulse Amplitude Modulation (PAM) Block Diagram of PAM System PAM Block Diagram Description Block Diagram Description (cont. 1) Block Diagram Description (cont. 2) Block Diagram Description (cont. 3) Intersymbol Interference (ISI) Eye Diagrams Formula for Eye Closure Nyquist Criterion for No ISI Nyquists Criterion (cont.) Raised Cosine Shaping Filters Raised Cosine Filters (cont.) Splitting the Shaping Interpolation Filter Bank Interpolation Filter Bank (cont.) Symbol Error Probability vs. SNR

Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide Slide

18 19 20 21 22 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40

Symbol Error Probability (2) Symbol Error Probability (3) Symbol Error Probability (4) Symbol Error Probability (5) Symbol Error Probability (6) Generating a Symbol Clock Tone Generating a Clock Tone (cont. 1) Generating a Clock Tone (cont. 2) Generating a Clock Tone (cont. 3) Generating a Clock Tone (cont. 4) Generating a Clock Tone (cont. 5) Generating a Clock Tone (cont. 6) Theoretical Exercises Theoretical Exercises (cont. 1) Theoretical Exercises (cont. 2) Theoretical Exercises (cont. 3) Theoretical Error Probability Making a PAM Signal Frequency Response of the Shaping Filter Using a Mailbox for Outputs Using a Mailbox (cont.) Generating a Baud Synch Signal Making a Clock Tone Generator Making a Tone Generator (cont.)

Slide Slide Slide Slide Slide Slide

40 41 42 43 44 45

Testing the Tone Generator Testing the Generator (cont. 1) Testing the Generator (cont. 2) Optional Team Exercise Optional Team Exercise (cont. 1) Optional Team Exercise (cont. 2)

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Chapter 11 Digital Data Transmission by Baseband Pulse Amplitude Modulation (PAM) Goals
Learn about baseband digital data transmission over bandlimited channels by PAM. Learn how to generate bandlimited PAM signals using baseband shaping lters realized by interpolation lter banks. Learn about intersymbol interference (ISI). Eye diagrams to show ISI The Nyquist criterion for no ISI Raised cosine shaping lters for no ISI Derive a symbol error probability formula. & Implement a symbol clock recovery method.

11-1

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Block Diagram of a Baseband PAM System

Serial to Parallel Converter
. . . .. .. .. .. ..

. .. .. .. .. ..

di . .

Map from J -bit Words to 2J Levels Transmitter

an

Impulse Modulator

s (t)

Transmit Filter GT (! )

s(t)

r(t)

Channel Frequency Response C (! )

v (t)

Additive Channel Noise Channel Model Receive Filter GR (! )

x(t)

Sampler and A/D

x(nT0

+ )

Adaptive Equalizer

y (nT )

- Quantizer

an ^

Symbol Clock Recovery ^ di Parallel to Serial Converter Receiver Map from 2J Levels to J -bit Words

. .. .. .. .. .. ..

. .. .. .. .. .. ..

&

11-2

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Description of the PAM Block Diagram

The transmitter input di is a serial binary data sequence with a bit rate of Rd bits/sec. Input bits are blocked into J-bit words by the serial-to-parallel converter. Input blocks are mapped into the sequence of symbols an which are selected from an alphabet of M = 2J distinct voltage levels. For example, the following levels uniformly spaced by 2d are commonly used: i = d(2i 1) for i = M M + 1, . . . , 0, . . . , 2 2

The minimum level is (M 1)d and the maximum level is (M 1)d. The symbol rate is fs = 1/T = Rd /J symbols/sec or baud. &

11-3

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PAM Figure Description (cont. 1)

The impulse modulator output is

s (t) =
k=

ak (t kT )

The bandlimiting transmit lter output is

s(t) =
k=

ak gT (t kT )

The combination of impulse modulator and transmit lter is a mathematical model for a D/A converter followed by a lowpass lter. The channel is modeled as a lter C() followed by an additive noise source. The receive lter eliminates out-of-band noise and, in conjunction with the transmit lter, forms a properly shaped pulse. &

11-4

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PAM Figure Description (cont. 2)

The combined transmit lter, channel, and receive lter frequency response is G() = GT ()C()GR ()

and the corresponding impulse response is g(t) = gT (t) c(t) gR (t) = F 1 {G()} The combined lter G() is called the baseband shaping lter. The output of the receive lter is

x(t) =
k=

ak g(t kT ) + v(t) gR (t)

&

The output of the receive lter is sampled at a rate that is an integer multiple N of the symbol rate fs . Typically, N might be 3 or 4. These samples are used by the Symbol Clock Recovery system to lock the receiver symbol clock to the transmitter clock.
11-5

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PAM Figure Description (cont. 3)

The Adaptive Equalizer is an FIR lter with adjustable taps that automatically compensates for channel amplitude and phase distortion. It also corrects for small deviations in the transmit and receive lter responses from their ideal nominal values. A least mean-square error (LMS) adaptation algorithm is used most often. The equalizer output is sampled at the symbol rate and quantized to the nearest ideal level. The Quantizer output is mapped to the corresponding J-bit binary word and converted back to a serial output data sequence. &

11-6

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Baseband Shaping and Intersymbol Interference (ISI)

An impulse response with the following property is said to have no intersymbol interference: 1 for n = 0 g(nT ) = [n] = 0 otherwise

Then, if the additive noise is zero, the samples of the receive lter output at the symbol instants are x(nT ) =
k=

ak [n k] = an

which are exactly the transmitted symbols. The transmit and received lters are usually designed so their cascade, GT ()GR (), has no ISI with, perhaps, some compromise equalization for the expected channel. &

11-7

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2

Eye Diagrams

1.5

0.5

0.5

1.5

0.2

0.4

0.6

0.8 1 1.2 Normalized Time t/T

1.4

1.6

1.8

Eye Diagram for a Raised Cosine Channel with 12% Excess Bandwidth With zero noise, baud rate samples of the receive lter output are

x(nT ) =
k=

ak g(nT kT )

&

11-8

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x(nT ) = g(0) an +

k= k=n

g(nT kT ) ak g(0)

The right-hand sum is the ISI for the received symbol. The worst case ISI occurs when the symbols ak have their maximum magnitude (M 1)d and the same sign as g(nT kT ). Then

(M 1)d

k= k=n

g(nT kT ) g(0) g(kT ) g(0)

(M 1)d

k= k=0

The peak fractional eye closure is dened to be D = = (M 1) d

k= k=0

g(kT ) g(0)

When is less than 1, the eyes are open. &

11-9

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The Nyquist Criterion for No ISI

The impulse response samples are 1 g(nT ) = 2

G()ejnT d

Let s = 2fs = 2/T . Then

g(nT ) =
k=
s 2

1 s

s 2

ks

s ks 2

1 G()ejnT d T

=
k=

1 s

s 2

1 G( ks )ej(ks )nT d T

Recognizing that ejkns T = ekn2 = 1 and taking the sum inside the integral gives 1 g(nT ) = s &
s 2

G ()ejnT d
s 2

11-10

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Nyquists Criterion (cont.)

where 1 G () = T

k=

G( ks )

The function G () is called the aliased or folded spectrum. There is no ISI if and only if G () = 1 because then the integral at the bottom of the previous slide becomes 1 g(nT ) = s
s 2

ejnT d = [n]
s 2

&

11-11

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Raised Cosine Baseband Shaping Filters

response of a raised cosine lter is for || (1 ) s 2
T 2

[0, 1] is the excess bandwidth factor. 1 0.8 0.6 G() 0.4 0.2 00 0.2 0.4 0.6 f /f s 0.8 1

The frequency T T 2 G() = 0

for (1 ) s || (1 + ) s 2 2 elsewhere

1 sin

||

s 2

&

Example for T = 1 and = 0.5

11-12

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Raised Cosine Filters (cont.)

sin
s 2 t

The corresponding impulse response is g(t) =

s 2 t

cos s t 2 1 4(t/T )2

The program C:\DIGFIL\RASCOS.EXE computes samples of the impulse response modied by the Hamming window. Special Cases If = 0, the raised cosine lter becomes an ideal at lowpass lter with cuto frequency s /2. If = 1, the frequency response has no at region and is one cycle of a cosine function raised up so it becomes 0 at the cuto frequency of s . As the bandwidth is increased by making closer to 1, the impulse response decays more rapidly.

&

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Splitting the Shaping Between the Transmit and Receive Filters

If the channel amplitude response is at across the signal passband and the noise is white, the amplitude response of the combined baseband shaping lter should be equally split between the transmit and receive lters to maximize the output signal-to-noise ratio, that is, |GT ()| = |GR ()| = |G()|1/2 Their phases can be arbitrary as long as the combined phase is linear. When raised cosine shaping is used, the transmit and receive lters are called square-root of raised cosine lters. C:\DIGFIL\SQRTRACO.EXE can be used to compute the impulse response of a square-root of raised cosine lter. &

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Implementing the Transmit Filter by an Interpolation Filter Bank

This approach places the computational burden on the DSP and allows a simple analog output lowpass lter to be used. The transmit lter continuous-time output is

s(t) =
k=

ak gT (t kT )

Let t = nT + m(T /L) to get

s nT + m T = L
k=

ak gT nT + m T kT L for m = 0, 1, . . . , L 1

Now let L discrete-time interpolation sublters be dened as gT,m (n) = gT (nT + m T ) for m = 0, . . . , L 1 L

&

11-15

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Interpolation Filters (cont.)

Notice that the sublters, gT,m (n), are FIR lters with T spaced taps. The L output samples for the symbol period starting at time nT are

s(nT + m T ) = L
k=

ak gT,m (n k)

for m = 0, 1, . . . , L 1 Next, the sublter outputs are multiplexed to a D/A converter at the rate of Lfs samples/second. Finally, the D/A output is passed through a simple analog lowpass lter. The interpolation lter bank is illustrated in the next slide. In practice, the transmit lter impulse response is truncated to a nite duration by a window function.

&

11-16

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An Interpolation Filter Bank

-

T;

(n) (n)

B B B

s(nT )
B B B

T;

..
-

Q Q

D/A +
L L

Lowpass Filter

s(t)
-

T ;L

(n)

s nT

Symbol Error Probability vs. SNR

Assumptions: The frequency response of the channel is a constant over the signal bandwidth. Symbols from the M level alphabet are used with equal probability. Symbols selected at dierent times are uncorrelated random variables. The additive noise is white and Gaussian with two-sided power spectral density N0 /2.

&

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Symbol Error Probability (2)

The combined baseband shaping lter has a raised cosine response with excess bandwidth factor and the shaping is split equally between the transmit and receive lters. Therefore, the transmit and receive lters both have square-root of raised cosine responses. It can be shown [Lucky, Salz, and Weldon, pp. 52-53] that the average transmitted power is E{a2 } 1 n Ps = T 2

|GT ()|2 d

With the square-root of raised cosine transmit lter, this reduces to E{a2 } n Ps = T &

11-18

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Symbol Error Probability (3)

With equally likely levels, the expected squared symbol value is a =

2

E{a2 } n

2 = M

M/2

k=1

d2 [d(2k 1)] = (M 1) 3
2 2

and the transmitted power is d2 Ps = (M 2 1) 3T The noise at the output of the square-root of raised cosine receive lter has the variance 1 2 = 2

(1)

N0 N0 |GR ()|2 d = 2 2

(2)

Also, the channel noise power in the Nyquist band (s /2, s /2) is PN & 1 = 2
s /2 s /2

N0 N0 d = 2 2T

(3)

11-19

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Symbol Error Probability (4)

Since there is no ISI, the samples of the receive lter output have the form x(nT ) = an + vR (nT ) where vR (nT ) is a sample of the channel noise ltered by the receive lter. The received sample x(nT ) is quantized to the nearest ideal level an . Error Probability for Inner Points For the M 2 inner levels, an error is made if the noise magnitude exceeds d, half the distance between points. In this case, the symbol error probability is PI = P (|vR (nT )| > d) = 2Q(d/) where Q(x) is the Gaussian tail probability

Q(x) =
x

1 t2 e 2 dt 2

&

11-20

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Symbol Error Probability (5)

which is accurately approximated for x > 2 by
2 1 x Q(x) e 2 x 2

Error Probability for the Outer Points For the level (M 1)d the error probability is PO+ = P (vR (nT ) < d) = Q(d/) For the outer level (M 1)d the error probability is PO = P (vR (nT )) > d) = Q(d/) = PO+ The total symbol error probability is Pe = = & 1 1 M 2 PI + PO+ + PO M M M M 1 2 Q(d/) M

11-21

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Symbol Error Probability (6)

Solving the transmitted power equation (1) for d and using formula (2) for the receive lter output noise power and formula (3) for the channel noise power, the error probability can be expressed in terms of the channel signal-to-noise ratio Ps /PN as 1/2 M 1 Ps 3 Pe = 2 Q M M 2 1 PN

Symbol Clock Recovery

The receiver must lock its local symbol clock frequency and phase to those in the received signal. A method, good for bandlimited systems, for deriving the symbol clock from the received signal is illustrated below.
x(t) Pre lter - B (! ) q (t)

- Squarer

p(t)

Bandpass Filter q 2 (t) H (! )

z (t)

PhaseLocked Loop

( - t)

&

System to Generate a Symbol Clock Tone

11-22

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Generating a Clock Tone (cont. 1)

The receive lter output x(t) is rst passed through a prelter with frequency response B(). This is a bandpass lter centered at s /2, half the symbol frequency.
B()

s /2 0 s /2

Let the combined baseband shaping lter and prelter frequency and impulse responses be G1 () = G()B() and g1 (t) = g(t) b(t) Then, the prelter output is

q(t) = &
k=

ak g1 (t kT )

11-23

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Generating a Clock Tone (cont. 2)

The prelter output is squared to get
p(t) = q 2 (t)

=
k= m=

ak am g1 (t kT )g1 (t mT )

p(t) is passed through a bandpass lter H() centered at the symbol rate s .
H()

s 0 s

&

z(t) looks like a sinusoid at the clock frequency with slowly varying amplitude and phase. Its zero crossings cluster together. z(t) is applied to a phase-locked loop to generate a stable symbol clock.

11-24

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Generating a Clock Tone (cont. 3)

It will be assumed that the symbols are a sequence of zero-mean uncorrelated random variables. Therefore, E{ak am } = a2 k,m The expected value of the squarer output is

E{p(t)} =
k= m=

E{ak am }g1 (t kT )g1 (t mT )

This reduces to

(t) = E{p(t)} = a2
k=

2 g1 (t kT )

(t) is periodic with period equal to the symbol period T . It can be expressed as a Fourier series of the form

E{p(t)} =
k=

pk ejks t

&

11-25

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Generating a Clock Tone (cont. 4)

1 pk = T a2 T
T

E{p(t)}ejks t dt
0

It can be shown that

2 2 g1 (t)ejks t dt

pk

= =

a T 2

G1 ()G1 (ks ) d

The expected value of the output bandpass lter is also periodic with the Fourier series expansion

E{z(t)} =
k=

zk ejks t

where zk = = & pk H(ks ) a2 H(ks ) T 2

G1 ()G1 (ks ) d

11-26

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Generating a Clock Tone (cont. 5)

By selecting the prelter B() to be a narrow band lter that passes components only near s /2, it can be seen that pk = 0 except for k = 1, 0, or 1. By selecting the output bandpass lter H() so that it only passes spectral components near s , the k = 0 term is removed and only the symbol frequency components for k = 1 remain. When the baseband shaping lter has zero excess bandwidth, that is, when G() = 0 for || s /2, all the Fourier coecients pk are zero for k = 0 since the nonzero portions of G() and G(ks ) do not overlap. This timing recovery method then fails. &

11-27

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Generating a Clock Tone (cont. 6) Condition for Good Timing Recovery

When G1 () is symmetric about s /2 and is bandlimited to the interval s /4 < || < 3s /4 and H() is symmetric about s , it can be shown that the variance of z(t) is zero and perfect timing recovery is possible. When these symmetry conditions are nearly met, the variations in the zero crossings of the timing wave z(t) are very small and the receiver can track the symbol clock frequency by locking to the zero crossings.

&

11-28

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Theoretical Exercises

1. Write a C function to generate pseudo-random four-level symbols. The function should use a 23-stage self synchronizing shift register sequence generator with the connection polynomial h(D) = 1 + D18 + D23 or 1 + D5 + D23

as discussed in Chapter 9, to generate the binary sequence dn . Generate the four-level sequence an from pairs of binary symbols (d2n , d2n+1 ) according to the rule an = (1)d2n (1 + 2d2n+1 )d where d is a desired scale factor. 2. Draw a vertical axis and show the 4 levels. Label each level with its analog value and also the corresponding pair of binary digits.

&

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Theoretical Exercises (cont. 1)

3. Design an interpolation lter bank. Use the program C:\DIGFIL\RASCOS.EXE to generate the sublter coecients. Use a symbol rate of fs = 1/T = 4 kHz. First, use an excess bandwidth factor of = 1.0.

Truncate the shaping lter impulse response to the interval [4T, 4T ] with a Hamming window. Generate L=16 samples of the PAM signal per symbol interval, that is, generate the sequence s(kT /16). 4. Generate data for an eye diagram that extends over two symbol intervals. Write enough pairs (mod(k, 32), s(kT /16)) to a le to form a reasonably lled out 4-level eye diagram. &
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Theoretical Exercises (cont. 2)

The function, mod(k, 32), is the remainder when k is divided by 32 and ranges from 0 to 31. As k increases, mod(k, 32) cycles throught the values 0, 1, . . . , 31. This performs the function of resetting the trace to the left-hand side every two symbols. When k reaches a multiple of 32 and mod(k, 32) = 0, you should write the three extra points (32, s(kT /16)), (32, 0) and (0, 0) to the le before writing (0, s(kT /16). These three points accomplish the following: (32, s(kT /16)) continues the trace to the right edge of the plot. (32, 0) moves the trace vertically to 0. (0, 0) moves the trace horizontally at 0 from the right edge back to the origin. Otherwise, retrace lines will be drawn &

11-31

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Theoretical Exercises (cont. 3)

through the eye diagram for some plotting programs.

5. Plot the shaping lter impulse response and eye diagram for = 1. 6. Change to 0.125 and generate a new eye diagram. Plot the new shaping lter impulse response and eye diagram. Discuss dierences in the impulse responses and eye diagrams for the two cases. 7. Comment on how the excess bandwidth factor aects the required symbol sampling time accuracy in the receiver. 8. Square-Root of Raised Cosine Shaping Repeat the raised cosine exercises for a square-root of raised cosine baseband shaping lter, but only for = 0.125. Also, compute the peak fractional eye closure dened on Slide 11-9 from the shaping lter impulse response. &
11-32

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9. Theoretical Error Probability The error probability can be expressed in terms of the channel signal-to-noise ratio Ps /PN as M 1 Pe = 2 Q M 3 Ps M 2 1 PN
1/2

under the following conditions: A at channel frequency response. Symbols from the M level alphabet are used with equal probability. Symbols selected at dierent times are uncorrelated random variables. The noise is white and Gaussian with power spectrum N0 /2. The transmit and receive lters have square-root of raised cosine responses. Plot Pe for M = 2, 4, and 8 on the same graph as a function of the channel signal-to-noise ratio Ps /PN . Plot Pe on a logarithmic scale and 10 log10 (Ps /PN ) dB on a linear scale.

&

11-33

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Making a PAM Signal with the DSK

Generate a four-level PAM signal using a raised cosine baseband shaping lter with = 0.125. Truncate the shaping lter impulse response to the interval [4T, 4T ] by a Hamming window. Use symbol rate fs = 4 kHz. Generate L = 4 PAM signal samples per symbol with an interpolation lter bank. The D/A sampling rate should be set to 4 4 = 16 kHz. Use the same 23-stage scrambler you used for the theortical exercises. Write the output samples to the left channel. You may write your program entirely in C or in combined C and assembly. Use -o3 optimization. &

11-34

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Frequency Response of the Shaping Filter

Let the impulse response of the baseband shaping lter, viewed as an FIR lter with T /4 tap spacing, be g(nT /4) for n = 16, 15, . . . , 16 gn = 0 elsewhere
16

The frequency response of this lter is G() =

n=16 16

gn ejnT /4

g0 + 2
n=1

gn cos(nT /4)

Compute and plot the amplitude response of your lter in dB over the frequency range of 0 to 8 kHz. &

11-35

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Using a Mailbox to Store and Output Samples

There are a variety of ways to structure the program. Here is one approach to try. Write output samples to the McBSP DXR with an interrupt routine triggered by the serial port transmit interrupts (XINT). Write the samples to the left channel. Determine the symbol timing by counting interrupts modulo 4. Set up an 8-word circular buer as a mail box. One half of the buer (4 words) will be used to hold the output samples for the current symbol period, and the remaining half will be used to store the four samples for the next symbol period. Each symbol period, the input and output halves will be swapped. These are sometimes called ping pong buers.
Output 0 1 2 3 0 1 2 3

&

Input

11-36

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Using a Mailbox (cont.)

Initialize the output pointer to the address of the rst word in the buer and the input pointer to fth word. Before starting data transmission, set the interrupt count to 0 to indicate the start of a symbol. At the start of each symbol period, generate four output samples, and write them to the mailbox. Do this in the main routine. The input pointer should be incremented circularly after each sample is written to the mailbox. After the four samples are written to the mailbox, the main routine should wait for the interrupt count to become 0. Remember, the interrupt routine should count interrupts modulo 4.

&

11-37

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The transmit interrupt service routine should write the sample addressed by the output pointer to the DXR, increment the output pointer circularly modulo 8, and increment the interrupt count modulo 4.

Generating a Baud Synch Signal

Test your program by observing the eye diagram on the oscilloscope. Use DC coupling for the scope probe. You will need a signal to synchronize the sweeps with the symbol period to get a display like your theoretical plot. One way to generate a synch signal is to create a 4000 Hz square-wave on the right channel codec output. You can do this by putting an integer like A = 16000 in the lower half of the word sent to the codec for rst two samples in a baud and A in the second two samples. Explain why the output does not look like a square wave. &
11-38

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Making a Clock Tone Generator

Write a program for the DSP to implement the symbol clock recovery system discussed in Slides 11-22 11-28 to the point labelled z(t) on Slide 11-22. Do not implement the phase-locked loop. Use the same raised cosine baseband shaping lter you designed for the C6x PAM signal generator. The sampling rate for all operations in the symbol clock tone generator should be 4fs = 16 kHz. Write the PAM output samples to the left channel D/A converter. Connect the left channel line output to the left channel line input. Send the input samples to your clock tone generation function. For the prelter, B(), design a second-order IIR lter with a center frequency of fs /2 = 2 kHz and roughly a 100 Hz 3 dB bandwidth. &
11-39

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Making a Tone Generator (cont.)

For the postlter, H(), design a second-order bandpass IIR lter with a center frequency of 4 kHz and a 3 dB bandwidth of roughly 25 Hz. You should experiment with these bandwidths and observe how they aect the system performance. Write the tone generator system output samples z(nT /4) to the right channel output.

Testing the Clock Tone Generator

1. First drive your baseband shaping lter with the alternating two-level symbol sequence an = (1)n d. This is called a dotting sequence in the modem jargon. Send the shaping lter output samples to the left channel output and the clock tone generator output samples to the right channel output. &
11-40

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Testing the Tone Generator (cont. 1)

Observe the left and right channel outputs simultaneously on the oscilloscope. Notice that an = (1)n = cos s nT which 2 are symbol rate samples of a cosine wave that has a frequency of half the symbol rate. The sampled signal has spectral components at the set of frequencies {s /2 + ks ; k = , . . . , } The shaping lter will pass the 2 kHz component and heavily attenuate the other components in the [0, 8) kHz band. In other words, the PAM signal should be very close to a 2 kHz sine wave. &

11-41

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Testing the Tone Generator (cont. 2)

Check that the tone generator output is a 4 kHz sine wave locked to the PAM signal. 2. Next, use a two-level pseudo-random symbol sequence having values d. Use the shift register generator to select the levels. Observe the PAM signal and clock tone generator output on the oscilloscope. They should still be locked together. Comment on how the tone generator output looks compared to the output with the dotting sequence. 3. Finally, use the full four-level pseudo-random input symbol sequence. Observe the output of the clock recovery system on the oscilloscope and compare it with the previous cases. &

11-42

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Optional Team Exercise

If you are interested in doing more with PAM, team up with an adjacent group. Make one setup a PAM transmitter and the other a PAM receiver. Transmit a two-level PAM signal. The levels should be selected by the output of the 23-stage scrambler with an input of 0. Use a 4 kHz symbol rate. Sample the received signal at 16 kHz. Even though the transmitter and receiver both use a sampling frequency of 16 kHz, there will be slight dierences due to small physical and temperature dierences in the oscillator crystals and circuit components. You will have to devise a method for synchronizing the symbol clock in the receiver to the symbol clock in the transmitter. &

11-43

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Optional Team Exercise (cont. 1)

The sampling phase of the codec cannot be altered, so you will have to pass the received samples through a variable phase interpolator that compensates for the phase dierence between the transmit and receive clocks. Variable phase interpolators are discussed in the next chapter. You can lock the phase of the receiver symbol clock to the positive zero crossings of the symbol clock tone generator. In addition you will have to compensate for any delays in the system so that samples are taken at the symbol instants, that is, at the point where the eye has its maximum opening. You should also process the received samples with a baseband version of the T /2 spaced adaptive equalizer described in Chapter 15. The equalizer will automatically adjust for any symbol sampling phase oset but not for a frequency oset. &
11-44

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Optional Team Exercise (cont. 2)

If the received signal has little distortion (inter-symbol interference), you can add a lter at the transmitter output or receiver input to simulate channel distortion. Then you can observe how the equalizer converges and opens the eye. Quantize the selected symbol rate samples to a binary sequence. Descramble this sequence and check that the output is all 0s. Add Gaussian noise to the received samples in the DSP and make a plot of the bit-error rate vs. SNR.

&

11-45