الفصل 3 - اتصالات رقمية

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Digital Communication ‫د محمود فرحان‬.

‫ ا‬: ‫مدرس المادة‬

Chapter Three
Source Coding Techniques

1- Introduction:
Analog waveform or signals are sampled into pulses. In digital pulses
modulation methods, the analog amplitude pulses are converted to digital
form. Thus each sample of the message signal is represented in binary (1, 0)
format.
2- Pulse Code Modulation:
The PCM technique samples the input signal x(t) at frequency 𝑓𝑠 ≥ 2𝑊. This
sampled pulse is then digitized by the analog to digital converter as shown in
Fig. 2-1.

Figure 2-1 the block diagram of PCM

The x(t) is bandlimited to W by LPF. The sample and hold circuit then samples this
signal at the rate above of Nyquist rate 𝑓𝑠 ≥ 𝑊. The sampled signal 𝑥(𝑛𝑇𝑠 )is discrete
in time and continuous in amplitude. The quantizer is convert it to q discrete level
by rounding each sample to fixed digital level with minimum error (quantization
error). The input to the quantizer 𝑥(𝑛𝑇𝑠 ) (for example) can take any values between
(−4𝛿 𝑡𝑜 + 4𝛿), the output of quantizer (𝑥𝑞 (𝑛𝑇𝑠 ) are available at
Digital Communication ‫د محمود فرحان‬.‫ ا‬: ‫مدرس المادة‬

𝛿 3𝛿 5𝛿 7𝛿
± ,± ,± 𝑎𝑛𝑑 ± as shown in Fig. 2-2. Thus the maximum quantization
2 2 2 2
𝛿
error is ± .
2

The quantized signal (𝑥𝑞 (𝑛𝑇𝑠 ) is converted by encoder to v digits binary word, and
then converted to serial bit stream to generate single baseband signal as shown in
Fig. 2-3.

2.1 Transmission bandwidth in PCM:


Each quantized sample can be represent
by v digits: 𝑞 = 2𝑣 , where q is the total
number of digital levels.
The number of sample is 𝑓𝑠 , and each
sample represent by v bits then:
Signaling rate of PCM: 𝑟 = 𝑣 × 𝑓𝑠
And we have 𝑓𝑠 ≥ 2𝑊.
The bandwidth of PCM given half
1
signaling rate: 𝐵𝑟 ≥ 𝑟
2
1
𝐵𝑟 ≥ 𝑣𝑓𝑠 Figure 2-2:(a) Transfer characteristic of a quantizer
2
(b) Variation of quantizer error

Since 𝑓𝑠 ≥ 2𝑊

∴ 𝐵𝑟 ≥ 𝑣𝑊

Figure 2-3: quantizing of sample value


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2-2 PCM Receiver:

Figure 2-4 shows the block diagram of PCM receiver. The regenerator is to reshapes
the pulse and removes the noise. The signal is then converted into parallel digital
words for each sample.

Figure 2-4 PCM receiver

The digital word is converted to its analog value 𝑥𝑞 (𝑡) along with sample and hold
(S/H), then passed through lowpass reconstruction filter to get 𝑦𝐷 (𝑡). There is
quantization error between reconstructed signal 𝑥(𝑘𝑇𝑠 ) and original signal 𝑥(𝑡) as
shown in figure 2-5. This can reduced by increasing bits ‘v’, but this increases the
bandwidth.

Figure 2-5: Reconstructed waveform


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2-3 Quantization Noise in PCM:


The quantization error is expressed as:

𝜀 = 𝑥𝑞 (𝑛𝑇𝑠 ) + 𝑥(𝑛𝑇𝑠 )

The range of amplitude of input 𝑥(𝑛𝑇𝑠 ) is −𝑥𝑚𝑎𝑥 𝑡𝑜 + 𝑥𝑚𝑎𝑥 and it is mapped into
q levels. So that total amplitude range 2𝑥𝑚𝑎𝑥 is divided into q levels with step size
𝛿.

2𝑥𝑚𝑎𝑥
𝛿=
𝑞

𝛿 𝛿
We have the maximum quantization error is ± , or 𝜀𝑚𝑎𝑥 = | |
2 2

The mean square value of quantization error is:

𝛿 𝛿
2 2 𝛿
3 2
1 1 𝜀
𝐸 (𝜀 2 ) = ∫ 𝜀 2 𝑓𝜀 (𝜀)𝑑𝜀 = ∫ 𝜀 2 𝑑𝜀 = [ ]
𝛿 𝛿 3 −𝛿
𝛿 𝛿 2
−2 −2

1 3 1 3
2
1 (𝛿 ) ( )
𝛿 1 𝛿3 𝛿3 𝛿2
𝐸 (𝜀 ) = [ + ]= [ + ]=
𝛿 3 3 3𝛿 8 8 12

2
𝑉𝑛𝑜𝑖𝑠𝑒
The noise power=
𝑅

2
𝑉𝑛𝑜𝑖𝑠𝑒
Assume R=1, then the noise power (normalized)=
1

2)
𝛿 2 /12 𝛿 2
𝐸 (𝜀 = =
1 12

The maximum signal power to quantization noise ratio:


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𝑆 𝑁𝑜𝑟𝑚𝑎𝑙𝑖𝑧𝑒𝑑 𝑠𝑖𝑔𝑛𝑎𝑙 𝑝𝑜𝑤𝑒𝑟


=
𝑁 𝛿 2 /12

We have 𝑞 = 2𝑣 , so:

2𝑥𝑚𝑎𝑥 2𝑥𝑚𝑎𝑥
𝛿= =
𝑞 2𝑣
𝑆
Substituting in above equation:
𝑁

𝑆 𝑁𝑜𝑟𝑚𝑎𝑙𝑖𝑧𝑒𝑑 𝑠𝑖𝑔𝑛𝑎𝑙 𝑝𝑜𝑤𝑒𝑟


= 2
𝑁 2𝑥
( 𝑚𝑎𝑥 )
2𝑣
12

Let normalized signal power as P

𝑆 𝑃 3𝑃
= 2 = 2
× 22𝑣
𝑁 2𝑥 (𝑥𝑚𝑎𝑥 )
( 𝑚𝑎𝑥
𝑣 ) /12
2

This equation shows that the signal to noise power ratio of quantizer increases
exponentially with increasing bits per sample. For normalized 𝑥𝑚𝑎𝑥

𝑆
= 3 × 22𝑣 × 𝑃
𝑁
𝑆 𝑆
( ) 𝑑𝐵 = 10𝑙𝑜𝑔10 ( ) 𝑑𝐵 = 10𝑙𝑜𝑔10 (3 × 22𝑣 ) = (4.8 + 6𝑣)𝑑𝐵
𝑁 𝑁

For normalized values of power the destination signal power ‘P’ is less than 1

So that

𝑆
( ) 𝑑𝐵 ≤ (4.8 + 6𝑣 )𝑑𝐵
𝑁
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Example 1:

A television signal with bandwidth of 4.2 MHz is transmitted using binary PCM.
The number of quantizatin levels is 512. Calculate:

i- Code word length


ii- Transmission bandwidth
iii- Final bit rate
iv- Output signal to quantization noise ratio

Solution:

i- 𝑞 = 2𝑣 → 512 = 2𝑣

𝑙𝑜𝑔512 = 𝑣 × 𝑙𝑜𝑔2 = 9 𝑏𝑖𝑡𝑠

Thus the code word length is 9 bits

ii- 𝐵𝑇 ≥ 𝑣𝑊 → 𝐵𝑇 ≥ 9 × 4.2 × 106


∴ 𝐵𝑇 ≥ 37.8 𝑀𝐻𝑧
iii- The signaling rate 𝑟 = 𝑣 × 𝑓𝑠 = 𝑣 × 2𝑊 = 9 × 2 × 4.2 × 106
𝑟 = 75.6 𝑀𝑏𝑝𝑠
1
Also we have 𝐵𝑇 ≥ 𝑟 𝑜𝑟 𝐵𝑇 ≥ 0.5 × 75.6 = 37.8 𝑀𝐻𝑧 which is same
2

value obtained earlier


𝑆
iv- ( ) 𝑑𝐵 = (4.8 + 6𝑣 )𝑑𝐵 = 4.8 + 6 × 9 = 58.8 𝑑𝐵
𝑁
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Example 2:

The bandwidth of signal input to the PCM is restricted to 4 kHz. The input varies
from -3.8V to +3.8 V and has the average power of 30 mW. The required signal
to noise ratio is 20 dB.

i- Calculate the number of bits required per sample.


ii- Outputs of 20 such PCM coder are time multiplexed. What is the mimum
required transmission bandwidth for the multiplexed signal?

Solution:

𝑆 𝑆
( ) 𝑑𝐵 = 10𝑙𝑜𝑔10 ( ) 𝑑𝐵 = 20𝑑𝐵
𝑁 𝑁

𝑆
∴ = 100
𝑁
𝑆 3𝑃 3×30×10−3 ×22𝑣
i- = (𝑥 2 )
× 22𝑣 → 100 = (3.8)2
𝑁 𝑚𝑎𝑥

1444
22𝑣 = = 24066.67
0.06
2𝑣𝑙𝑜𝑔2 = log(24066.67)
𝑣 ≅ 7 𝑏𝑖𝑡𝑠
ii- 𝐵𝑇 ≥ 𝑣𝑊 and for 20 multiplexed signals

𝐵𝑇 ≥ 20 × 7 × 4 𝑘𝐻𝑧 ≥ 840 𝑘𝐻𝑧

And signaling rate

𝑟 = 2𝐵𝑇 = 2 × 840 = 1680 𝑘𝑏𝑝𝑠


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Example 3:

The information in an analog signal voltage waveform is to be transmitted over


a PCM system with an accuracy of ∓0.1% (full scale). The analog signal has a
bandwidth of 100Hz and an amplitude range of -10 to +10 volts. Determine:

i- The number of levels required for such accuracy.


ii- The code word length.
iii- The minimum bit rate required.
iv- The bandwidth required for PCM signal.

Solution:

i- The maximum quantization error should be ∓0.1%, so:

𝜀𝑚𝑎𝑥 = ∓0.001

𝛿
But 𝜀𝑚𝑎𝑥 = | |
2

𝛿
| | = 0.001 → 𝑡ℎ𝑒 𝑠𝑡𝑒𝑝 𝑠𝑖𝑧𝑒 𝛿 = 0.002
2
2𝑥𝑚𝑎𝑥
We have 𝛿= , 𝑎𝑛𝑑 |𝑥𝑚𝑎𝑥 | = 10 𝑣𝑜𝑙𝑡𝑠
𝑞

2×10
Then 0.002 = → 𝑞 = 10000
𝑞

So that the number of levels are 10000.

ii- We have 𝑞 = 2𝑣 → 10000 = 2𝑣 → 𝑙𝑜𝑔10000 = 𝑣 𝑙𝑜𝑔2


∴ 𝑡ℎ𝑒 𝑐𝑜𝑑𝑒 𝑤𝑜𝑟𝑑 𝑙𝑒𝑛𝑔𝑡ℎ 𝑣 = 13.288 ≅ 14 𝑏𝑖𝑡𝑠
iii- We have the bit rat 𝑟 = 𝑣𝑓𝑠 = 𝑣 × 2 × 𝑊 = 14 × 2 × 100 = 2800 𝑏𝑝𝑠
1 1
iv- The bandwidth required 𝐵𝑇 ≥ 𝑟 ≥ × 2800 = 1400 𝐻𝑧
2 2
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H.W:

Q1/ The information in an analog waveform with maximum frequency 𝑓𝑚 = 3 𝑘𝐻𝑧


is to be transmitted over 16- levels PCM system. The quantization distortion is
specified not exceed 1% of peak to peak analog signal.

i- What is the number of bits per sample that should be used in


this PCM?
ii- What is minimum bit transmission rate?

Q2 / A signal of bandwidth 3.5 kHz is sampled, quantized and coded by PCM


system. The code signal is then transmitted over a transmission channel of
supporting a transmission rate of 50 kbps. Calculate the maximum signal to noise
that can obtained by this system. The input signal has peak to peak value of 4 volts
and rms value of 0.2 V.

Q3 / Consider an audio signal comprised of the sinusoidal term 𝑠(𝑡) =


3 cos(500𝜋𝑡 ).

i- Find the number of quantization level with an accuracy of 1%.


ii- Determine the signaling rate.
iii- The bandwidth of transmission channel.

2-4 Advantages of PCM:


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i- Effect of channel noise and interference is reduced.


ii- PCM permits regeneration of pulses along the transmission path. This
reduces noise interference.
iii- The bandwidth and signal to noise ratio are related by exponential law.
iv- Multiplexing of various PCM signals is easily possible.
v- Encryption or decryption can easily incorporated for security purpose.

2-5 Limitation of PCM:

i- PCM system are complex compared to analog pulse modulation method.


ii- The channel bandwidth is also increased because of digital coding of
analog pulses.

2-6 Modifications of PCM:

i- PCM can be modified to delta modulation. It is more simplified method of


implementation.
ii- The PCM can be used in wideband communications channels to overcome
the bandwidth problem.
iii- With the help of data comparison along with PCM, the redundancy can be
removed and data rate can be reduced.
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2-7 Differential Pulse Code Modulation (DPCM):

Any signal does not change fast, so that the value from present sample to next
sample does not differ by large amount. The adjacent samples of the signal carry
the same information with little difference as shown if figure 2-11.

Figure 2-6 Redundant information of PCM


It can be seen from figure 2-11 the samples 4𝑇𝑠 , 5𝑇𝑠 , 𝑎𝑛𝑑 6𝑇𝑠 are encoded to the
same value of (110). If this redundancy is reduced, the overall bit rate will decrease
and the number of bits required for one sample will also be reduced. This is called
Differential Pulse Code Modulation (DPCM).

DPCM works on the principle of prediction. The value of the present sample is
predicted from the past samples as shown in figure 2-12.

Figure 2-7 DPCM transmitter


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The comparator fined the difference between the actual sample value 𝑥 (𝑛𝑇𝑠 ) and
predicted signal 𝑥̂(𝑛𝑇𝑠 ) this is called error 𝑒(𝑛𝑇𝑠 ):

𝑒(𝑛𝑇𝑠 ) = 𝑥 (𝑛𝑇𝑠 ) − 𝑥̂(𝑛𝑇𝑠 )

This error will be quantized and encoded by small number of bits. Thus number of
bits per sample are reduced in DPCM. The quantization error can be written as:
𝑒𝑞 (𝑛𝑇𝑠 ) = 𝑒(𝑛𝑇𝑠 ) + 𝑞(𝑛𝑇𝑠 )
The prediction filter input is:
𝑥𝑞 (𝑛𝑇𝑠 ) = 𝑥̂ (𝑛𝑇𝑠 ) + 𝑒𝑞 (𝑛𝑇𝑠 )
Substituting by 𝑒𝑞 (𝑛𝑇𝑠 ) yields
𝑥𝑞 (𝑛𝑇𝑠 )=𝑥̂ (𝑛𝑇𝑠 ) + 𝑒 (𝑛𝑇𝑠 ) + 𝑞(𝑛𝑇𝑠 )
We have
𝑥 (𝑛𝑇𝑠 ) = 𝑒(𝑛𝑇𝑠 ) + 𝑥̂(𝑛𝑇𝑠 )

From the last equations;

𝑥𝑞 (𝑛𝑇𝑠 )=𝑥 (𝑛𝑇𝑠 ) + 𝑞(𝑛𝑇𝑠 )


To reconstruct the original signal at the receiver, the decoder first reconstructs the
quantized error signal as shown in figure 2-13.

Figure 2-8 Reconstruction of DPCM

The quantized error signals are summed up with prediction filter output to give the
quantized version of the original signal. The signal at the receiver differs from actual
signal by quantization error 𝑞(𝑛𝑇𝑠 ).
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2.8 Delta Modulation (DM):

DM transmit only one bit per sample. That is the present sample value is compared
with the previous sample of approximated signal which confined to two levels
(−𝛿 𝑎𝑛𝑑 + 𝛿). If the difference is negative ‘0’ bit is transmitted and ‘1’ bit is
transmitted for positive difference, as shown in 2-6.

Figure 9 Delta modulation waveform

- DM transmitter:
The block diagram of DM transmitter is shown in figure 2-7

Figure 10 DM transmitter
The error between sampled value of 𝑥(𝑡) and last approximated sample is
given by:
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𝑒(𝑘𝑇𝑠 ) = 𝑥 (𝑘𝑇𝑠 ) − 𝑥̂ (𝑘𝑇𝑠 )


From the waveform figure 𝑢(𝑘𝑇𝑠 ) is the present sample approximation of
staircase output and 𝑢[(𝑘 − 1)𝑇𝑠 ] = 𝑥̂ (𝑘𝑇𝑠 ) is the last sample approximation
of staircase output. Let the quantity of 𝑏(𝑘𝑇𝑠 ) be define as,
𝑏(𝑘𝑇𝑠 ) = 𝛿𝑠𝑖𝑔𝑛[𝑒(𝑘𝑇𝑠 )]
So that
𝑏(𝑘𝑇𝑠 ) = +𝛿 𝑖𝑓 𝑥(𝑘𝑇𝑠 ) ≥ 𝑥̂ (𝑘𝑇𝑠 ), 𝑏𝑖𝑛𝑎𝑟𝑦 ′1′ 𝑖𝑠 𝑡𝑟𝑎𝑛𝑠𝑚𝑖𝑡𝑡𝑒𝑑
𝑏(𝑘𝑇𝑠 ) = +𝛿 𝑖𝑓 𝑥(𝑘𝑇𝑠 ) < 𝑥̂ (𝑘𝑇𝑠 ), 𝑏𝑖𝑛𝑎𝑟𝑦 ′0′ 𝑖𝑠 𝑡𝑟𝑎𝑛𝑠𝑚𝑖𝑡𝑡𝑒𝑑
Where 𝑇𝑠 is sampling interval.
The summer of figure 2-7adds quantizer output (∓𝛿) with previous sample
approximation. This gives present sample approximation:
𝑢(𝑘𝑇𝑠 ) = 𝑢(𝑘𝑇𝑠 − 𝑇𝑠 ) + [∓𝛿 ] 𝑜𝑟
= 𝑢[(𝑘 − 1)𝑇𝑠 ] + 𝑏(𝑘𝑇𝑠 )
The previous sample approximation 𝑢[(𝑘 − 1)𝑇𝑠 ] is restored by delayed one
sample period 𝑇𝑠 . The sampled input signal 𝑥 (𝑘𝑇𝑠 ) and staircase
approximated signal 𝑥̂ (𝑘𝑇𝑠 ) are subtracted to get error signal 𝑒 (𝑘𝑇𝑠 ). Then
the one bit quantizer produces +𝛿 or – 𝛿 step size depending on the sign of
𝑒(𝑘𝑇𝑠 ), ‘1’ bit is transmitted for positive values and ‘0’ bit for negative values.
- DM receiver:
At the receiver shown in figure 2-8, the accumulator generates the staircase
approximated signal and is delayed by one sample period 𝑇𝑠 . It adds +𝛿 step
to the previous sample if the input bit is ‘1’ and subtract −𝛿 for ‘0’ input bit.
The low pass filter with cutoff frequency of highest frequency in 𝑥(𝑡).
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Figure 11 DM receiver
- Advantageous of DM:
i- DM transmit only one bit for one sample. Thus the signaling rate and
transmission channel bandwidth is quite small for DM.
ii- The DM transceiver system is very much simple.
- Disadvantageous of DM:

Figure 12 distortion of DM
i- Slop overload distortion: This distortion arises because of large dynamic
range of input signal. In this case the step size 𝛿 is too small for staircase
signal 𝑢(𝑡) to follow the steep segment of 𝑥(𝑡). Thus there is large error
between those signals. This error called slop overload distortion. To reduce
this error the step size should be increased when slop of signal 𝑥(𝑡) is high.
But since the step size is fixed it is called Linear Delta Modulation (LDM).
ii- Granular Noise (Hunting): It is occur when the step size is too large
compared to small variation in the input signal 𝑥(𝑡) which can be
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considered flat, while the staircase signal is oscillated by ±𝛿 around it. The
error in this case is called granular noise, so that step size should be small
to reduce this error.

2.9 Adaptive DM:


The large step size is required to reduce slope overload while small steps
are required to reduce granular noise. Adaptive DM shown in figure 2-8
is a modification of LDM to overcome these errors.

Figure 13 Transmitter of adaptive DM

The step size increases with steep segment of input signal and reduces
with small variation. This called Adaptive Delta Modulation (ADM).

At the receiver the logic for step size control is added is added as shown
in figure 2-9.

Figure 14 ADM receiver


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If one bit quantizer output is high ‘1’ the step size may be doubled for next
sample and vice versa as shown in figure 2-10.

Figure 14 Waveform of ADM


The previous input and the present input decided the step size.

2- Intersymbol Interference (ISI):


ISI arises because dispersive nature of the communication channel, thus the errors
are introduced in the detected data at the receiver. The binary data can be transmitted
in baseband or passband. One of the baseband system of digital data is PAM which
is have only two amplitude level corresponding to binary "1" and "0". Successive
binary digits can combined into symbol. Line codes generate discrete PAM signals
which transmitted in baseband form (without any modulation) over the channel. Fig.
5 shows the block diagram of such baseband transmission system.

Figure 15
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The binary data (bk) is applied to the data encoder which generate the pulse
waveform x(t) represented mathematically by:

𝑥 (𝑡) = ∑ 𝐴𝑘 𝑔(𝑡 − 𝑘𝑇𝑏 )


𝑘=−∞

𝑇𝑏 is the duration of each input binary bit.


𝑔(𝑡) is the shaping pulse.
+𝑎 𝑖𝑓 𝑏𝑘 = 1
And 𝐴𝑘 = {
−𝑎 𝑖𝑓 𝑏𝑘 = 0
The signal x(t) is then passed through the transmitting filter which combine all the
necessary transmitting circuits and systems with transfer function of HT(f). The
signal is then passed through the channel having transfer function HC(f). The channel
delivers the signal to the receiving filter which combine all the necessary receiving
circuits and systems with transfer function of HR(f). The output of the receiving filter
is y(t) is a noisy replica of the transmitted signal x(t). The signal y(t) is then sampled
synchronously with clock pulse at the transmitter having sampling instants 𝑡 = 𝑖𝑇𝑏 .
The sampled signal y(ti) is then given to decision device which compare the input
signal with threshold "𝜆":
If y(ti) ˃ 𝜆 select symbol '1'
If y(ti) ≤ 𝜆 select symbol '0'
ISI Problem: Consider the output y(t) of the receiving filter represented by:

𝑦(𝑡) = µ ∑ 𝐴𝑘 𝑝(𝑡 − 𝑘𝑇𝑏 )


𝑘=−∞

Here µ is scaling factor and P(t) is the different from that of g(t)
We have 𝑡 = 𝑖𝑇𝑏 then:

𝑦(𝑡) = µ ∑ 𝐴𝑘 𝑝(𝑖𝑇𝑏 − 𝑘𝑇𝑏 )


𝑘=−∞

𝑦(𝑡) = µ ∑ 𝐴𝑘 𝑝(𝑖 − 𝑘)𝑇𝑏


𝑘=−∞
Digital Communication ‫د محمود فرحان‬.‫ ا‬: ‫مدرس المادة‬

Let us rearrange above equation:


𝑦(𝑡) = µ 𝐴𝑖 𝑝(0) + µ ∑ 𝐴𝑘 𝑝(𝑖 − 𝑘 )𝑇𝑏


𝑘=−∞
𝑘≠𝑖

𝑎𝑛𝑑 𝑖 = 0, ∓1, ∓2, ∓3, … … ….


If ISI is absent then the second term of above equation will not be present, and p(t)
is normalized such that 𝑝(0) = 0 yelds: 𝑦(𝑡) = µ 𝐴𝑖 . At 𝑡 = 𝑖𝑇𝑏 , the correct bit is
𝐴𝑖 . Observe that it is decoded correctly in absence of ISI, but it is not possible to
eliminate the second term totally. The ISI can be reduced by proper design of
transmitted filter HT(f), receive filter HR(f), and channel filter HC(f).

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