CHETTINAD COLLEGE OF ENGINEERING & TECHNOLOGY

NH-67, TRICHY MAIN ROAD, PULIYUR, C.F. – 639 114, KARUR DT

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

COURSE MATERIAL

Subject Name: Analog and Digital communication Class / SEM: B.E. (CSE) / III Subject Code: CS2204 Staff Name: Suganya. J

UNIT III

DIGITAL TRANSMISSION Syllabus Introduction, Pulse modulation, PCM – PCM sampling, sampling rate, signal to

quantization noise rate, companding – analog and digital – percentage error, delta

modulation, adaptive delta modulation, differential pulse code modulation, pulse

transmission – Inter symbol interference, eye patterns.

Objectives

• Define digital transmission

• Study pulse modulation and methods of pulse modulation

• Define and describe pulse code modulation and its derivatives

• Define SNR, ISI and eye patterns

Introduction

Digital Transmission is the transmittal of digital signals between two or more points in a communication system.

The signals can be binary or any other form of discrete-level digital pulses. The

original source information may be in digital form, or it could be analog signals converted

to digital pulses prior to transmission and converted back to analog signals in receiver.

With digital transmission systems, a physical facility such as a pair of wires, coaxial

cable or an optical fiber cable is required to interconnect the various points within the

system.

The pulses are contained in and propagated down the cable. Digital pulses cannot

be propagated through a wireless transmission system such as earth’s atmosphere or free

space (vacuum).

Advantages of Digital Transmission

1. The primary advantage is noise immunity. Digital signals are inherently less susceptible to interference caused by noise, since it is not necessary to evaluate the precise amplitude, phase of frequency to ascertain its logic condition. Instead, pulses are evaluated during a precise time interval and a simple determination is made whether the pulse is above or below a prescribed reference level.

2. Digital signals are best suited for processing and combining using a technique

called multiplexing.

3. Digital signal processing (DSP) is the processing of analog signals using digital methods and includes band-limiting the signals with filters, amplitude equalization and phase shifting.

4. It is simpler to store digital signals than analog signals and the transmission rate of

digital signals can be easily changed to adapt to different environments and to interface with different types of equipment.

5. Digital transmission systems are more resistant to analog systems to additive noise.

Analog systems use signal amplification and the noise produced in electronic circuits is additive (i.e. it accumulates). Therefore SNR deteriorates each time an analog signal is amplified. Also since there are number of circuits the analog signal is passed through in long distance communication, it limits the total distance of transmission. In digital system, signal regeneration is used instead of amplification. Digital regenerators sample the noisy signal and then reproduce an entirely new digital signal with same SNR as original transmitted signal. Therefore digital signals can be transported for longer distances than analog signal.

6. The digital signals are simpler to measure and evaluate.

7. It is easier to compare the error performance of one digital system to another

digital system. 8. The transmission errors can be detected and corrected more easily and more

accurately than it is possible with analog signal. Disadvantages of Digital Transmission

1. The transmission of digitally encoded analog signal requires significantly more

bandwidth than simply transmitting the original analog signal.

2. Analog signals must be converted to digital pulses prior to transmission and

converted back to their original analog form at the receiver, thus necessitating the

additional encoding and decoding circuitry.

Pulse Modulation

Pulse modulation consists essentially of sampling analog information signal,

converting those samples into discrete pulses and transporting those pulses from source to

destination over a physical transmission medium.

There are 4 predominant methods of pulse modulation: 1) Pulse Width Modulation

2) Pulse Position Modulation 3) Pulse Amplitude Modulation 4) Pulse Code Modulation. Pulse Width Modulation

It is also called as pulse length modulation or pulse duration modulation.

In PWM, the width of a constant amplitude pulse is varied proportional to the amplitude of

the analog signal at the time the signal is sampled. The maximum analog signal amplitude produces the widest pulse and the minimum analog

signal amplitude produces the narrowest pulse. But all pulses have the same amplitude.

Pulse Position Modulation

In PPM, the position of a constant width pulse within a prescribed time slot is

varied according to the amplitude of the sample of analog signal. If the amplitude of the analog signal increases, the pulses are positioned farther to the right

within the prescribed time slot. Therefore, the highest amplitude sample produces a pulse to far right and lowest

amplitude sample produces a pulse to far left.

Pulse Amplitude Modulation The amplitude of a constant width constant position pulse is varied according to the

amplitude of the sample of analog signal. PAM waveforms resemble the original analog signal more than the waveforms for PWM

or PPM.

Diagram showing different methods of Pulse Modulation

Applications of Pulse Modulation

1. PAM is used as an intermediate form of modulation with PSK, QAM and PCM although it is seldom used by itself.

2. PWM and PPM are used in special purpose communication systems mainly for military but seldom used for commercial digital transmission systems.

Pulse Code Modulation

PCM is the only digitally encoded digital modulation techniques commonly used for

digital transmission.

With PCM, the pulses are of fixed length and fixed amplitude.

PCM is a binary system where the presence of a pulse represents logic 1 condition and

the absence of a pulse represents a logic 0 condition.

The band-pass filter limits the frequency of the analog input signal to the standard

voice-band frequency range of 300 Hz to 3000 Hz. The sample-and-hold circuit periodically samples the analog input signal and converts

those samples into a multilevel PAM signal.

The analog-to-digital converter (ADC) converts the PAM samples to parallel

PCM codes, which are converted to serial binary data in the parallel-to-serial

converter and then outputted onto the transmission line as digital pulses.

The transmission line repeaters (regenerative repeaters) are placed at

prescribed distances to regenerate the digital pulses.

In the receiver, the serial-to-parallel converter converts serial pulses received

from the transmission lines to parallel PCM codes.

The digital-to-analog converter (DAC) converts the parallel PCM codes to

multilevel PAM signals.

The hold circuit is basically a low pass filter that converts the PAM signals back

to its original analog form.

An integrated circuit (IC) that performs the PCM encoding and decoding

functions is called a codec (coder/decoder).

PCM Sampling

The function of a sampling circuit in a PCM transmitter is to periodically sample the

continually changing analog input voltage and convert those samples to a series of

constant amplitude pulses that can more easily be converted to binary PCM code.

For the ADC to accurately convert a voltage to a binary code, the voltage must be

relatively constant so that the ADC can complete the conversion before the voltage

level changes.

If not, the ADC would be continually attempting to follow the changes and may

never stabilize on any PCM code.

There are two basic techniques to perform the sampling function: Natural

Sampling and Flat-top Sampling.

Natural sampling is shown in Figure. Natural sampling is when the tops of the

sample pulses retain their natural shape during the sample interval. It is difficult for

an ADC to convert the sample to a PCM code in natural sampling.

With natural sampling, the frequency spectrum of the sampled output is different

from that of an ideal sample.

The amplitude of the frequency components decreases for higher harmonics in a

(Sin x)/x manner. This alters the frequency spectrum requiring the use of

frequency equalizers (compensation filters) before recovery by a low pass filter.

The most common method used for sampling voice signals in PCM system is flat-top

sampling. This is accomplished by a sample-and-hold circuit. The purpose of a sample-and-hold circuit is to periodically sample the continually

changing analog input voltage and convert those samples into a series of constant

amplitude PAM voltage levels. With flat-top sampling, the input voltage is sampled with narrow pulses and then held

relatively constant until the next sample is taken. Below figure shows the flat-top sampling. The sampling process alters the frequency

spectrum and introduces an error called aperture error, which occurs, when the amplitude

of the sampled signal changes during the sample pulse time. Due to this, it is not possible

for the recovery circuit in the PCM receiver to exactly reproduce the original analog signal

voltage. The magnitude of the error depends on how much the analog signal voltage changes while

the sample is being taken and the width of the sample pulse.

Flat-top sampling introduces less aperture distortion than natural sampling and

operates with a slower ADC.

Figure shows the schematic diagram of a sample-and-hold circuit. Its operation is

explained as follows: The field-effect transistor Q1 acts as a sample circuit. When the circuit is turned ON, Q1

provides a low impedance path to deposit the analog sample voltage across the

capacitor C1. The time that the Q1 is ON is called the aperture or acquisition time. Essentially, the C1 is the hold circuit. When Q1 is OFF, C1 does not have a complete path

to discharge through and so it stores the sampled voltage. The storage time of the

capacitor is called the A/D conversion time. During this time, the ADC converts the

sample voltage to a PCM code. The acquisition time should be very short such that a minimum change occurs in the

analog signal while it is deposited across C1. If the input to ADC is changing while performing the conversion, aperture distortion

results. By having a short aperture time and keeping the input to the ADC relatively constant,

the sample-and-hold circuit can reduce the aperture distortion.

The output impedance of the voltage follower Z1 and the ON resistance of Q1 should be as

small as possible. This ensures that the RC charging time constant of the capacitor is kept very

short allowing the capacitor to charge or discharge rapidly during the short acquisition time. The rapid drop in the capacitor voltage immediately following each sample is due to the

redistribution of the charge across C1. The inter-electrode capacitance between the gate and drain of the FET is placed in series with

C1. When the FET is OFF, it acts as a capacitive voltage divider network. The gradual discharge across the capacitor during the conversion time is

called the droop. This is caused by the capacitor discharging through its own leakage

resistance and the input impedance of the voltage follower Z2.

To prevent this, the input impedance of Z2 and the leakage resistance of C1 should be as high as

possible.

The voltage follower Z1 and Z2 isolate the sample-and-hold circuit (Q1 and C1) from the

input and output circuitry.

Problem

For the sample-and-hold circuit, determine the largest value capacitor that can be used.

Use output impedance for Z1 of 10Ω, an ON resistance for Q1 of 10Ω, an acquisition time of

10µs, a maximum peak-to-peak input voltage of 10V, a maximum output current from Z1 of

10mA and an accuracy of 1%.

Sampling Rate

Nyquist sampling theorem establishes the minimum sampling rate (fs) that can be used

for a given PCM system

For a sample to be reproduced accurately in a PCM receiver, each cycle of the analog

input signal (fa) must be sampled at least twice. The minimum sampling rate is equal to twice the highest audio input frequency

(i.e. fs = 2fa). If fs < 2fa, an impairment called alias or fold over distortion occurs. The minimum Nyquist sampling rate is fs = 2 fa where fs is the minimum

Nyquist sample rate (Hz) and fa is the maximum analog input frequency (Hz).

A sample-and-hold circuit is a non-linear device (mixer) with two inputs; the sampling

pulse and the analog input signal. Non-linear mixing (heterodyning) occurs between

these two signals.

Signal-to-quantization noise ratio

The 3-bit PCM coding scheme consists of linear codes which means that the magnitude

change between any two successive codes is the same. The magnitude of their

quantization error is also the same.

The maximum quantization noise is half the resolution (quantum).

The worst possible signal-to-quantization noise voltage ratio (SQR) occurs when the

input signal is at its minimum amplitude (101 or 001).

Mathematically, the worst case SQR is

SQR= (resolution)/Qe

= Vlsb / (Vlsb /2)

= 2

The worst case (minimum) SQR occurs for the lowest magnitude quantization Voltage (±1V). The minimum SQR is

SQR (min) = 1/ 0.5 = 2

or in dB, SQR(min) = 20 Log 2 = 6 dB. For a maximum amplitude input signal of 3V (either 111 or 011), the maximum quantization noise is

also equal to the resolution divided by 2. SQR for a maximum input signal is

or in dB, SQR(max)=20 Log 6 = 15.6 dB

Though the magnitude of the quantization error remains constant throughout the

entire PCM code, the percentage error is not constant. The percentage error decreases

as the magnitude of the sample increases.

For comparison purposes and to illustrate SQR is not constant throughout the entire

range of sample amplitudes, the expression of SQR is for voltage values.

The analog input waveform varies in magnitude. Therefore, the SQR is not constant.

The quantization error or distortion caused by digitizing an analog sample is expressed

as an average signal power-to-average noise power ratio.

For linear PCM codes, the signal power to quantization noise power ratio is

determined by

SQR (dB) = 10 Log [(v2/R)/ (q2/12)/R]

Where R = Resistance (ohms)

v = rms signal voltage (volts)

q = quantization interval (volts)

v2/R = average signal power (watts)

(q2/12)/R = average quantization noise power (watts)

If the resistances are assumed to be equal,

SQR= 10 Log [v2/ (q2/12)]

= 10.8+ 20 log (v/q) Companding

Companding is the process of compressing and then expanding.

With companded systems, the higher amplitude analog signals are compressed prior to

transmission and then expanded in the receiver.

Companding is a means of improving the dynamic range of a communications system.

Analog companding

Analog compression is implemented using specially designed diodes inserted

in analog signal path in a PCM transmitter prior to sample-and-hold circuit.

Analog expansion can also be implemented using diodes placed just after low

pass filter in PCM receiver.

In transmitter, the dynamic range of analog signal is compressed, sampled and

then converted to a linear PCM code.

In receiver, PCM code is converted to a PAM signal, filtered and then

expanded back to its original dynamic range.

The different signal distributions require different companding characteristics.

For e.g., the voice quality telephone signal requires a relatively constant SQR

performance over a wide dynamic range which means that the distortion must

be proportional to signal amplitude for all input signal levels. It requires a log

compression ratio, which requires an infinite dynamic range and infinite

number of PCM codes. It is impossible to achieve.

There are two methods of analog companding currently used that closely

approximates a logarithmic function and often called Logarithmic PCM codes.

µ-law companding

Where V max - maximum uncompressed analog input voltage (volts) Vin– amplitude of input signal at a particular instant of time (volts)

µ - parameter used to define the amount of compression (unit less) Vout – compressed output voltage (volts) Below figure shows the compression curves for several values of µ. If µ increases, the

compression also increases. For µ = 0, the curve is linear since no compression. The parameter µ determines the range of signal power in which SQR is relatively

constant.

The voice transmission requires a minimum dynamic range of 40 dB and a 7-bit PCM

code. For a relatively constant SQR and a 40 dB dynamic range, a µ ≥ 100 is required.

A - Law Companding

Established in Europe by ITU-T, used to approximate true logarithmic

companding.

For an intended dynamic range, A-law has a slightly flatter SQR than µ-Law.

A-law is inferior to µ-law in terms of small signal quality (idle channel noise).

Compression characteristics for A-law companding is,

Digital Companding Digital companding involves compression in transmitter after samples has been

converted to a linear PCM code and expansion in receiver prior to PCM decoding.

Analog signal is first sampled and converted to a linear PCM code and then linear code

is digitally compressed. In receiver, the compressed PCM code is expanded and then decoded to convert back

to analog.

The most recent digitally companded PCM system uses 12-bit linear PCM code and 8-

bit compressed PCM code.

Percentage error

It is also called as Digital Compression Error.

The magnitude of compression error is not same for all samples. The maximum

percentage error is same in each segment other than segments 0 and 1 since

there is no compression.

For comparison purposes, the percentage error introduced by digital

compression is,

% Error = 12 bit Encoded Voltage -12bitDecodedVoltage X100 12bitDecodedVoltage

There are several ways in 12-bti to 8-bit compression and 8-bit to 12-bit

expansion with hardware, the simplest and most economical method is with

look-up table in ROM.

Every function performed by PCM encoder and decoder is accomplished with a

single Integrated Chip (IC) called CODEC.

Most of the more recently codec are COMBO chips, including an anti-aliasing

band pass filter, sample-and-hold circuit, ADC in transmit section and DAC,

hold circuit and BPF in receiver section.

Delta Modulation

DM uses a single bit PCM code to achieve digital transmission of analog

signals.

With conventional PCM, each code is a binary representation of both sign and

magnitude of a particular sample. Therefore, multiple bit codes are required to

represent many values that the sample can.

Instead of transmitting the coded representation of the sample, only one bit is

transmitted which indicates whether the sample is larger or smaller than the

previous sample.

Algorithm for DM is quiet simple. If the current sample is less than the previous

sample, logic 0 is transmitted. If the current sample is larger than the previous

sample, logic 1 is transmitted. Delta Modulation Transmitter

The input analog signal is sampled and converted to a PAM signal which is compared

with the output of DAC. The output of DAC is a voltage equal to the regenerated magnitude of previous

sample, stored in up-down counter as a binary number. The up-down counter is increased or decreased depending on whether the previous

sample is larger or smaller than the current sample. The up-down counter is clocked at a rate equal to the sample rate. The up-down

counter is updated after each comparison. Initially the up-down counter is zeroed and

DAC is outputting 0 V. The first sample is taken, converted to a PAM signal and

compared with 0 V.

The output of the comparator is logic 1 condition (+1) indicating the current sample is

larger in amplitude than the previous sample.

On the next clock pulse, up-down counter is increased to a count of 1. DAC now

outputs a voltage equal to the magnitude of the minimum step size (resolution). The steps change the value at a rate equal to clock frequency (Sample rate). With the

input signal, the up-down counter follows the input analog signal and counts up until

the output of DAC exceeds the analog sample. Up-down counter begin counting down until the output of DAC drops below the

sample amplitude. In idealized situation, the DAC output follows the input signal. (As

Figure 4.17). Each time the up-down counter is incremented, logic 1 is transmitted and

each time it is decremented, logic 0 is transmitted.

Delta Modulation Receiver

The receiver is identical to the transmitter except for comparator. As logic 1s and 0s are received, up-down counter is incremented or decremented

accordingly. The output of DAC in decoder is identical to the output of DAC in transmitter. With DM, each sample requires the transmission of only one bit. The bit rate

associated with DM is lower than the conventional PCM system.

There are two problems associated with DM that don’t occur with conventional PCM.

They are slope overload and granular noise.

Slope Overload

Slope overload occurs when the analog input signal changes at a faster rate

than the DAC can maintain.

The slope of analog signal is greater than the delta modulator can maintain and

is called slope overload.

Increase in clock frequency decreases the probability of slope overload

occurring.

Granular Noise

When the original analog input signal has relatively constant amplitude, the

reconstructed signal has variations that were not present in the original signal,

so called granular noise.

Granular noise is analogous to quantization noise in conventional PCM. It is

reduced by decreasing the step size.

To reduce the granular noise, a small resolution is needed. To reduce the

possibility of slope overload occurring, a large resolution is needed. So a

compromise is necessary.

Granular noise is more prevalent in analog signal that have gradual slopes and

whose amplitudes vary only a small amount.

Slope overload is more prevalent in analog signal that have steep slopes and

whose amplitudes vary rapidly.

Adaptive Delta Modulation

Adaptive delta modulation is a delta modulation in which the step size of DAC is

automatically varied depending on the amplitude characteristics of the analog input

signal.

When the output of a transmitter is a string of consecutive 1’s or 0’s, the slope of the

DAC output is less than the slope of analog signal in either positive or negative

direction.

DAC lost the exact track of analog samples and the possibility of occurrence of slope

overloading is high.

With ADM, after a predetermined number of consecutive 1’s or 0’s, step size is

automatically increased.

After the next sample, if DAC output amplitude is still below the sample amplitude,

next step is increased even further until DAC catches up the analog signal.

When an alternative sequence of 1’s and 0’s is occurring, the possibility of granular

noise is high.

DAC automatically reverses to its minimum step size and reduce the magnitude of

noise error.

The common algorithm is: When 3 consecutive 1’s or 0’s occur, step size of DAC is

increased or decreased by a factor of 1.5.

Various other algorithms may also be used for adaptive delta modulators depending on

particular system requirements.

Differential Pulse Code Modulation

In a typical PCM encoded speech waveform, there are often successive samples taken

in which there is a little difference between the amplitudes of two samples.

It is necessary to transmit several identical PCM codes which are redundant.

DPCM is designed to take advantages of sample to sample redundancies in typical

speech waveform. DPCM transmitter is shown in Figure 4.22.

Here, the difference in amplitude of 2 successive samples is transmitted rather than the

actual sample.

Since the range of sample difference is typically less than the range of

individual, fewer bits are required.

The analog input signal is band limited to one half the sample rate compared with the

preceding accumulated signal level in differentiator.

The output of the differentiator is the difference between the two signals. The

difference is PCM encoded and transmitted.

ADC operates the same as in conventional PCM system except that it typically uses

fewer bits per sample.

Each received sample is converted back to analog, stored and then summed with the

next sample received.

Integration is performed on the analog signal though it can also be performed digitally.

Pulse Transmission

All digital carrier systems involve the transmission of pulses through a

medium with a finite bandwidth.

Highly selective systems require a large number of filter sections, but it is impossible to

implement practically.

Practical digital systems use filters with a bandwidth that are approximately 30% or

more in excess of ideal Nyquist bandwidth.

Figure shows the output waveform from a band limited communication channel when

a narrow pulse is applied to its input. The band limiting of a pulse causes the energy

from the pulse to be spread over a significantly larger time in the form of secondary

lobes (Ringing tails).

The output frequency spectrum corresponding to a rectangular pulse is

referred as (Sin x)/x response, f(ω) = (T) sin(ωT/2)/ (ωT/2)

Where ω=2πf (radian) and T= Pulse width (Sec).

Figure (b) shows the distribution of total spectrum power. Approximately 90% of signal

power is contained within the first spectral null. (f = 1/T). The signal confined to a bandwidth of B = 1/T and pass most of the energy from original

waveform.

The amplitude at the middle of each pulse interval needs to be preserved. If the bandwidth is confined to B = 1/2T, the maximum signaling rate achievable through a LPF

with a specified bandwidth without causing excessive distortion is given as Nyquist rate

(Mathematically R =2B where R is the signaling rate given as 1/T and B is the specified

bandwidth).

Inter Symbol Interference

The input signal is a random binary NRZ sequence. The output signal reaches its

full value for each transmitted pulse at the center of each sampling interval.

At sampling instants, the signal does not attain always the maximum value. Ringing

tails of several pulses have overlapped interfering with the major pulse lobe.

When there is no time delay, the energy in the form of spurious responses from

one pulse appears during the sampling instant of another pulse. This phenomenon

is called Inter-Symbol Interference (ISI).

ISI is important in the transmission of pulses over circuits with a limited bandwidth

and linear phase response.

Rectangular pulse won’t remain as rectangular in less than infinite bandwidth.

Narrower the bandwidth, more rounded the pulses. If phase distortion is

excessive, the pulses tilt and affect the next pulse.

When pulses from more than one source are multiplexed together, the amplitude,

frequency and phase responses become critical. ISI causes crosstalk between channels that occupy adjacent time slots in a TDM

carrier system. Special filters called equalizers are inserted in the transmission path

to equalize the distortion for all frequencies creating uniform transmission medium

reducing transmission impairments. The four primary causes of ISI are:

Timing Inaccuracies In digital transmission system, the transmitter timing inaccuracies cause ISI, if the rate

of transmission does not conform to the ringing frequencies designed into the

communication channel. The timing inaccuracies of this type are insignificant. Since the receiver clocking

information is derived from the received signal (with noise), inaccurate sampling

timing mostly occur in receivers than in transmitters. Insufficient Bandwidth Timing errors are less likely to occur if the transmission rate is well below the channel

bandwidth. (Nyquist bandwidth < Channel bandwidth)

When channel bandwidth reduces, ringing frequency also reduces which results in more ISI.

Amplitude Distortion Filters are placed in the communication channel to band limit the signal and reduce the

predicted noise and interference. They are also used to produce a specific pulse

response. The frequency response of a channel cannot be predicted absolutely. When the frequency characteristics of a communication channel depart from the

predicted values, pulse distortion occurs. Pulse distortion results when the peaks of

pulses are reduced causing improper ringing frequency in time domain. The compensation for amplitude/pulse distortion is amplitude equalization. Phase Distortion Each pulse is simply the superposition of a series of harmonically related sine waves

with specific amplitude and phase relationships. Phase distortion occurs when the

relative phase relations of the individual sine waves are altered. Phase distortion also occurs when the frequency components undergo different

amounts of time delay while propagating through the transmission medium. Special delay equalizers are placed in the transmission path to compensate the varying

delays thus reducing the phase distortion. Phase equalizers can be manually adjusted

or designed to automatically adjust themselves to varying transmission characteristics.

Eye Pattern

The performance of a digital transmission system depends on the ability of a

repeater to regenerate the original pulses.

The quality of regeneration process depends on the decision circuit within the

repeater and quality of the signal at the input to the decision circuit.

The performance of digital transmission system is measured by displaying received

signal on an oscilloscope and triggering the time base at the data rate. All the

waveform combinations are superimposed over adjacent signaling intervals. Such a

display is called Eye pattern or Eye diagram.

Eye pattern is a convenient technique to determine the effects of degradations

introduced into the pulses as they travel to the regenerator.

Figure shows the test set up to display an Eye pattern. The received pulse stream is

fed to the vertical input of the oscilloscope and symbol clock is fed to the external

trigger input while sweep rate is set approximately equal to symbol rate.

Eye pattern is generated by a symmetrical waveform for ternary signal in which the

individual pulses at the input to the regenerator have a cosine-squared shape.

In an m-level system, there are m-1 separate eyes. The vertical lines labeled +1, 0

and -1 correspond to ideal received amplitudes. The horizontal lines are separated

by a signaling interval T corresponds to ideal decision times. Vertical hairs

represent decision time and horizontal hairs represent decision levels.

Eye pattern shows the quality of shaping and timing. It also discloses any noise and errors

that is present in line equalization. Eye opening defines a boundary within which no

waveform trajectories can exist under any code pattern condition. To generate pulse sequence without errors, the eyes must be open and decision cross hairs

must be within open area. The pulse degradation reduces the size of ideal eye. At the

center of the eye, opening is 90% indicating only minor ISI degradation due to filtering

imperfections. Small degradation is due to the non-ideal Nyquist amplitude phase characteristics of

transmission system.

ISI degradation is ISI = 20Log h where H = ideal vertical opening (cm)

H

And h = degraded vertical opening (cm). Here ISI = 20 Log (90/100) = 0.915 dB. The overlapping signals pattern does not cross horizontal zero line at exact integer

multiples of symbol clocks. This impairment is called data transmission jitter. This jitter has

an effect on symbol timing clock recovery circuit. Excessive effect degrades the

performance of regenerative sections.

1 MARK

1. In PCM, what follows the input band pass filter? a. ADC b. Sample-and-hold c. Low-pass filter d. DAC

2. During times when there is no analog input signal, the only input to the PAM is random, thermal noise called a. Overload distortion b. Idle channel noise c. Slope overload d. Quantization error

3. Results if the magnitude of the sample exceeds the highest quantization interval. a. Overload distortion b. Idle channel noise c. Slope overload d. Quantization error

4. An error which results from the rounding the magnitude of the sample to the nearest valid code. a. Overload distortion b. Idle channel noise c. Slope overload d. Quantization error

5. Results when the input signal frequency is higher than the DAC can handle. a. Overload distortion b. Idle channel noise c. Slope overload d. Quantization error

6. A problem associated with delta modulation. a. Granular noise b. Overload distortion c. Quantization noise d. None of the above

7. A type of sampling where the sample voltage is held at a constant amplitude during the A/D conversion time; this is done by sampling the analog signal for a short period of time. a. Natural sampling b. Constant sampling c. ADC sampling d. Flat-top sampling

8. A type of sampling, the sample time is made longer and the analog-to-digital conversion takes place with changing analog signal. This introduces more aperture distortion and requires a faster A/D converter? a. Natural sampling b. Constant sampling c. ADC sampling d. Flat-top sampling

9. If the sampling frequency is less than twice the maximum analog input frequency, which of

the following will occur. a. Aliasing b. Fold over distortion c. Crossover distortion d. A and B above

10. Magnitude of the minimum step size.

a. Quantization b. Resolution c. Pixel d. All of the above

2 MARKS

1. Define digital transmission.

2. What are the advantages and dis-advantages of digital transmission?

3. What are the four most common methods of pulse code modulation?

4. Define aperture and acquisition time.

5. What is the difference between natural and flat-top sampling?

6. Define companding.

7. Define slope overload noise.

8. Define granular noise.

9. What is Nyquist sampling rate?

10. Write the expression for A-Law and µ -law companding.

11. Define percentage error.

DETAIL

1. Explain the companding techniques for analog and digital in detail with block diagram.

2. Explain the PCM Transmission in detail

3. Explain delta modulation PCM in detail

4. Explain Intersymbol interference in detail

5. Explain adaptive delta modulation in detail

6. Explain Eye pattern in detail