EXP.NO:

DATE

STUDY OF LINE CODING AND DECODING TECHNIQUES

AIM:

To study the following Line Coding techniques and to obtain their waveforms using digital trainer kit.

1. Non-return to zero – LEVEL coding (NRZ – L)

2. Non-return to zero –MARK coding (NRZ – M)

3. Non-return to zero – SPACE coding (NRZ-S)

4. Unipolar Return to zero coding (URZ)

5. Biphase –Level ( Manchester Coding )

6. Bi-phase-Mark

7. Bi-phase-Space

8. Return to zero-Alternate Mark Inversion Coding

DEVICES, COMPONENTS AND EQUIPMENT REQUIRED:

SL.No NAME TYPE/RANGE QUANTITY

1

2

3

Line coding Trainer Kit

Cathode Ray Oscilloscope

Patch cords

DCLT 005 & 006

30MHz

-

1

1

(10-15)

THEORY:

Line coding consists of representing the digital signal to be transported by an amplitude- and time-discrete signal that is optimally tuned for the specific properties of the physical channel (and of the receiving equipment). The waveform pattern of voltage or current used to represent the 1s and 0s of a digital data on a transmission link is called line encoding.

NRZ-L:

Non return to zero level is used extensively in digital logic circuits. A binary one is represented by one voltage level and a binary zero is represented by another voltage level.

NRZ-M:

With Non return to zero Mark, the one, or mark, is represented by a change in level, and the zero, or space, is represented by no change in level. This is often referred to as differential coding. NRZ-M is used primarily in magnetic tape recording.

NRZ-S:

Non return to zero space is the complement of NRZ-M. A one is represented by no change in level, and a zero is represented by a change in level.

UNI-RZ:

With Unipolar return to zero, a one is represented by a half-bit-wide pulse, and a zero is represented by the absence of a pulse.

Bi-phase-Level: (Manchester)

Bi-phase-level is better known as Manchester coding. With bi-Φ-L, a one is represented by a half bit wide pulse positioned during the first half of the bit interval and a zero is represented by a half bit wide pulse positioned during the second half of the bit interval.

Bi-phase-Mark:

With Bi-Φ-M, a transition occurs at the beginning of every bit interval. A one is represented by a second transition one bit half interval later; a zero is represented by no second transition.

Bi-phase-S:

With Bi-Φ-S, a transition also occurs at the beginning of every bit interval. A one is represented by no second transition: a zero is represented by a second transition one bit half interval later.

RZ-AMI:

Return to zero- Alternate mark inversion is a signalling scheme used in telephone systems. The ones are represented by equal amplitude alternating pulses. The zeros are represented by the absence of pulses.

PROCEDURE:

1. Set up the DCLT 005 in standalone mode.

2. Connect the unit with the given power supply.

3. Connect s-clock to the coding clock.

4. Connect s-data to the input data.

5. Connect one channel of the scope to the s-data and the data to the coded data to be observed.

6. Power on the unit.

7. Repeat the connections for channel decoding.

8. Set DCLT-005 in conjunction with DCLT-006 and power on the unit with the supply.

9. Connect the data output (BS2) of DCLT-005 to the B2 pin of DCLT-006 and make other connections as shown in Figure to obtain the decoded data pattern.

RESULT:

MODEL WAVEFORMS

Figure 1.1 Waveforms of different Line Coding Techniques

LINE CODING SIMULATION:

RETURN TO ZERO-AMI:

UNIPOLAR RETURN TO ZERO:

LAYOUT DIAGRAM OF LINE ENCODER AND DECODER:

1. What are the factors to be considered before selecting a line code for a certain application? List the applications of line codes.

Various factors that are considered for selecting a line encoding format are:1. Transmission bandwidth2. Power efficiency3. Error detection and correction capability4. Favorable power spectral density5. Adequate timing content6. Transparency.

The line-coded signal can directly be put on a transmission line, in the form of variations of the voltage or current (often

using differential signaling).

The line-coded signal (the "baseband signal") undergoes further pulse shaping (to reduce its frequency bandwidth) and

then modulated (to shift its frequency) to create an "RF signal" that can be sent through free space.

The line-coded signal can be used to turn on and off a light source in free-space optical communication, most commonly used in

an infrared remote control.

The line-coded signal can be printed on paper to create a bar code.

The line-coded signal can be converted to magnetized spots on a hard drive or tape drive.

The line-coded signal can be converted to pits on an optical disc.

2. What is clock recovery? Which line coding schemes possess this property? Explain how clock signal can be recovered from coded data.

Some digital data streams, especially high-speed serial data streams (such as the raw stream of data from the magnetic head of a disk drive) are sent without an accompanying clock signal. The receiver generates a clock from an approximate frequency reference, and then phase-aligns to the transitions in the data stream with a phase-locked loop (PLL). This process is commonly known as recovery. Manchester coding, 8B/10B, 64B/66B are some line coding schemes that possess the property of clock recovery. Clock recovery is accomplished with the help of phase locked loop. COSTAS loop is also used for clock recovery.

3. List the common transmission line impairments. Why is normalized bandwidth ( BW time product ) preferred compared to Bandwidth while comparing line coding schemes?Cross talk, Echo or Signal Return, Frequency shift, Non Linear Distortion, Jitter, Transients are some common transmission line impairments. The time-bandwidth product is invariant to the following operations: Shifting waveform in time Shifting waveform in frequency(modulation in time) Multiplying the function by a constant Scaling of independent variableThe time-bandwidth product is thus a robust measure of combined time and frequency spreadof a signal. It is essentially a property of the shape of the waveform.Hence normalized bandwidth is preferred compared to bandwidth while comparing line coding schemes

4. What is m-ary line coding

In M-ary coding, M bits at a time (M = 2k) are take and a waveform (or symbol) is created

EXP NO:

DATE:

PULSE CODE MODULATION

AIM:

To analyze and study the modulation and demodulation of Pulse Code Modulation technique.

DEVICES, COMPONENTS AND EQUIPMENT REQUIRED:

SL.NO NAME TYPE/RANGE QUANTITY

1.

2.

3.

4.

PCM transmitter kit

PCM Receiver kit

Patch cords

Dual time CRO

DCLT-003

DCLT-004

-

(0-30)MHz

1

1

10(approx.)

1

SPECIFICATIONS OF DIGITAL TRAINER KIT:

Input channel: Two channels which are time division multiplexed and pulse code modulated.

On board signals: Two variable amplitude sine waves of 1 kHz and 500 Hz

Frame synchronization Technique: Phase locked loop technique.

Power supply: 230V, 50 Hz.

THEORY:

Alex H.Reeves is credited with inventing pulse code modulation in 1937 at its Paris laboratories. Pulse code modulation system is the only digitally encoded modulation technique that is commonly used for digital transmission.PCM is a method of signal coding in which the message signal is sampled, the amplitude of each sample is rounded off to nearest one of a finite set of a discrete levels and encoded so that both time and amplitude are represents in discrete form. This allows the message to be transmitted by means of a digital waveform. This is a binary system where a pulse or a lack of pulse with in a prescribed time slot represents either logic 1 or a logic 0 condition. Pulse code modulation systems are complex in that the message signal is subjected to a large number of operations. The essential operations in the transmitter of a pulse code modulation system are sampling, quantizing and encoding

The sampling, quantizing and encoding operations are usually performed in the same circuit, which is called an analog-to-digital convertor. Regeneration of improved signals occurs at intermediate points along the transmission path. At the receiver, the essential operation consists of one last stage of regeneration followed by decoding, then demodulation.

PCM TRANSMITTER:The band pass filter limits the frequency of analog input signal to the standard voice-band frequency range of 300Hz to 3000Hz.The sample and hold circuit periodically samples the analog input signal and converts those samples to a multilevel PAM signal. The analog-to-digital convertor (ADC) converts the PAM sample to parallel PCM codes, which are converted to a serial binary data in the parallel to serial convertor and then outputted on to the transmission line as serial digital pulses. The transmission line repeaters are placed at prescribed distances to regenerate the digital pulses. Refer fig 2.1

PCM RECEIVER:

In the receiver, the serial –to-parallel convertor converts serial pulses received from transmission line to parallel PCM codes. The digital-to-analog convertor (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 waveform. Refer fig 2.2

The use of digital representation of analog signals offers the following advantages:

Extremely noise resistant and hence mostly free from channel interference,

Efficient regeneration of the coded signal along the transmission path,

Moreover secrecy can be built in by proper coding.

It allows the following disadvantages

Large bandwidth requirement.

Synchronization is required between the transmitter and receiver.

BLOCK DIAGRAMS:

Figure 2.1 PCM TRANSMITTER

Figure 2.2 PCM RECEIVER

LAYOUT DIAGRAM:

Figure 2.3 Pulse Code Modulator and Demodulator trainer kit layout diagram

TABULATION:

SL.NO SIGNALS AMPLITUDE (Vp-p) FREQUENCY (Hz)

1

2

3

4

5

6

7

Input Signal-A

Input Signal-B

Sampling Clock Pulses A

Sampling Clock Pulses B

Modulated output Signal

Demodulated Signal A

Demodulated Signal B

8 Sampling rate = ( samples/second )

Number of bits per sample =

Bit Rate = number of multiplexed signal x sampling rate x bits per second =

Figure 2.4 Waveforms of Pulse Code Modulation

Figure 2.5 Waveforms of Pulse Code De modulator

PROCEDURE:

1. Connect the input sine wave of 500Hz to CH0 and 1 KHz to CH1 of the multiplexer.

2. Set up the speed selection switch to fast mode. Connect TXCLOCK from DCL-003 to RXCLOCK in DCL-004, connect TXSYNC from DCL- 003 to RXSYNC in DCL-004 and connect the PCM output from DCL-003 to RXDATA in DCL-004.

3. Set the error code selection switch to none as DCLT-003 and DCLT-004.

4. Observe the waveforms at DCLT-003 at the points TP5, TP6 for inputs TP7 and TP8 for sampling pulses;TP12 for TDM of samples of input signals.

5. Observe pulse code modulation output at TP48 or at PCM output of the pulse code modulation transmitter.

6. Connect the PCM output to the RXDATA of DCL-004.

7. Observe the sampling pulse at the receiver side at TP37 and TP36 and the demultiplexed channels of DCLT-005 at TP39 and TP40.

8. Also observe the reconstructed outputs at TP43 and TP44 for respective channel input.

RESULT:

LAYOUT DIAGRAM OF PULSE CODE MODULATOR AND DE MODULATOR TRAINER KIT:

1.State the Sampling theorem for low pass signals and the Nquist sampling rate.The sampling theorem requires that a lowpass signal be sampled at least at twice the highest frequency component of the analog band-limited signal. This in essence ensures that the spectral replicas that occur due to sampling do not overlap and the original signal can be reconstructed from the samples with theoretically no distortion.

The Nyquist rate is the minimum sampling rate required to avoid aliasing, equal to twice the highest frequency contained within the signal.

2.State the sampling theorem for bandpass signals.

In order for a faithful reproduction and reconstruction of a bandpass analog signal with bandwidth – Fb, the signal should be sampled at a Sampling frequency (Fs) that is greater than or equal to twice the maximum bandwidth of the signal.”

3.What is aliasing ?

In signal processing and related disciplines, aliasing refers to an effect that causes different signals to become indistinguishable (or aliases of one another) when sampled. It also refers to the distortion or artifact that results when the signal reconstructed from samples is different from the original continuous signal.

4.What is Quantisation noise?

Errors are introduced in the signal because of the quantization

process. This error is called "quantization error". We define the

quantization error as: ε= xq (nTs)- x(nTs)

5.Differentiate between linear and non linear quantisation.

In linear quantization every increment in the sampled value corresponds to a fixed size analogue increment. E.g. an 8 bit A-D or D-A with a 0 - 1 V analogue range has 1 / 256 = 3.9 mV per bit, regardless of the actual signal amplitude.

Non-linear quantization normally have some sort of logarithmic encoding (e.g. µ-Law or A-law), so that the increment for small sample values is much smaller than the increment for large sample values. Ideally the step size should be roughly proportional to the sample size. This translates to a fixed S/N ratio (due to quantization noise), regardless of the signal amplitude.

6.What is companding ? Why is it needed?

Companding refers to a technique for compressing and then expanding (or decompressing) an analog or digital signal. It is a combination of the words "compressing" and "expanding."

Companding is used in the digital systems primary in audio such as microphones (more effectively in wireless) to reduce the noise levels in the sound quality mainly owing to low-level radio frequency interference in the frequency channel. Companding reduces the noise and crosstalk levels at the receiver.

7.What is differential PCM?

Differential PCM is technique that codes the difference between sample points to compress the digital data. Because audio waves propagate in predictable patterns, DPCM predicts the next sample and codes the difference between the prediction and the actual point. The differences are smaller numbers than the numerical value of each sample on the full scale and thereby reduce the resulting bit stream

EXP.NO:

DATE:

DELTA MODULATION

AIM:

To construct a Delta Modulator and to study its performance.

DEVICES, COMPNONENTS AND EQUIPMENT REQUIRED:

SL.NO NAME TYPE/RANGE QUANTITY

1

2

3

4

5

6

7

Op-amp

Dual D-Flipflop

Resistors

Capacitors

Regulated Power Supply

Function Generator

Cathode Ray Oscilloscope

IC 741

IC 4013

10KΩ

4.7 KΩ

100 KΩ

0.01µF

(0-30)V

(0-1M)Hz

30MHz

3

1

3

1

2

2

1

1

1

THEORY:

Delta modulation 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 by a particular sample. Therefore the multiple bit code are required to represent to the values that the samples can be with delta modulation, rather than transmit a coded representation of samples. Gain for various messages is added to the points and the pulses are represented in the receiver by a suitable multiplexer.

Assume N similar message are multiplexed each being band limited with the maximum spectral extent of wf. At the transmitted side, the pulse modulator accepts an input gain of constant amplitude clock pulses. Modulator, according to a rate of ws/2 pulses per second, converts the pulses to an output, train of polar pulses with polarity determined by the error signal e(t). If e(t)is positive at the time of any given input pulses, the output pulse is positive, else it is negative. e(t)= f(t)-fq(t) Where f(t) = input message fq(t) = feedback signal Only a single bit is transmitted which simply indicate whether that sample is larger or smaller than the previous sample. The algorithm of a delta modulation system is quite simple. If the current sample is smaller than the previous sample the logical ’0’ is transmitted or else logical ‘1’ is transmitted. One bit coding is made possible by a feedback which is an integral part of the encoding process. The simplest form of delta modulation provides a staircase to the oversampled version of an input base band signal. In delta modulation the modulator transmit binary output pulses whose polarity depends on the difference between the modulating and feedback signal. These generated pulses having duration ‘T’ which is short relative to the time

Ts= 2/ws between the sample pulses. Thus the integrated output will be staircase type waveform with an up going step for a negative pulse. Each transmitted pulse is early encoded version of the signal e(t). The wave function of the loop is to force out the fq(t) to be an approximation of f(0). With delta modulation, each sample requires the transmission of any one bit. Therefore the bit rate association with delta modulation are lower than conventional PCM. However the slope overload of granular noise is the limitation of delta modulation.

∆=2δ ………(3.1)

Figure 3.1 Staircase approximation of Message signal

PROCEDURE:

1) Connections are given stage by stage as shown in figure 3.2.

2) The digital sampler is constructed first.

3) The message signal is applied to the non-inverting terminal of the comparator.

4) Without the feedback the waveforms are checked.

5) The sampling pulse is applied to the clock input of D-flipflop. Set the sampling frequency much greater than twice the message frequency.

6) The output of the sampler is given to the integrated circuit.

7) The output of the inverting amplifier is given as input to the inverting terminal of the comparator in the digital sampler.

8) With feedback the delta modulation output from the D-flipflop is observed.

RESULT:

:

TABULATION:

SL.NO SIGNALS AMPLITUDE(V (P−P)) FREQUENCY (Hz)

Bit Rate = Sample Rate =

1.What is the significance of delta modulation?

Delta modulation (DM or Δ-modulation) is an analog-to-digital and digital-to-analog signal conversion technique used for transmission of voice information where quality is not of primary importance. DM is the simplest form of differential pulse-code modulation (DPCM) where the difference between successive samples is encoded into n-bit data streams. In delta modulation, the transmitted data is reduced to a 1-bit data stream.

Its main features are:

The analog signal is approximated with a series of segments Each segment of the approximated signal is compared to the original analog wave to determine the

increase or decrease in relative amplitude The decision process for establishing the state of successive bits is determined by this comparison Only the change of information is sent, that is, only an increase or decrease of the signal amplitude from

the previous sample is sent whereas a no-change condition causes the modulated signal to remain at the same 0 or 1 state of the previous sample.

2.Write about slope overload error and granular error?

Slope Overload Distortion

Since DM depends on a discrete number of bytes to sample the audio information, the sampling rate imposes a limit on how closely it can approximate the shape of the audio wave. The "delta" is the size of the individual steps. Displayed graphically, the approximation of the sound input looks like a staircase. When the slope of the sound waveform exceeds the ability of DM's step size to keep up, this creates infidelity known as slope overload distortion. These distortions are common on steep, sinusoidal sections of a source wave.

Granular Noise Distortion

When the step size of a DM is too large in comparison to the input sound wave, a distortion is created. As DM seeks out a step that approximates the area, it selects levels that fluctuate both above and below the original form. This fluctuation is called granular noise. Granular distortions are especially common in flat or horizontal wave sections.

3.What is adaptive delta modulation?

There is a conflict between the requirements for minimization of slope overload and the granular noise. The one requires an increased step size, the other a reduced step size. There is a way to overcome this problem which adjusts the step size according to the slope of the signal being sampled. This is a variation of the basic delta modulation and is called the adaptive delta modulation.

EXP.NO:

DATE:

TIME DIVISION MULTIPLEXING-PAM

AIM:

To perform the time division multiplexing and demultiplexing of the four input signals and study the signals at various points at the transmitter and receiver.

DEVICES, COMPONENTS AND EQUIPMENT REQUIRED:

SL.NO NAME TYPE/RANGE QUANTITY

1

2

3

Digital Trainer Kit

Power supply

Patch Cords

DCLT-002

(0-30)V

-

1

1

(10-15)

TECHNICAL SPECIFICATIONS OF DCLT-002:

Input channels : 4

Switching : TDM

Modulation : PAM

Sampling rate : 32 KHz

Maximum signal bandwidth : 2 KHz

On-Board sine waves : 2KHz, 1KHz, 500Hz, 250Hz

Synchronization pulse generation : using variable dc level.

PLL : generates receiver clock and channel information.

Test point/Interconnection : 230V, 50Hz

THEORY:

Time Division Multiplexing is a type of multiplexing where two or more channels of information are transmitted over the same link by allocating different time interval for the transmission of each channel.

In PAM, the pulse is present for short duration and for most of the time between the two pulses, no signals are present. This free space between pulses can be occupied by pulses from other channels. So to make minimum utilization of transmission channel, TDM of PAM signal is done.

Each channel to be transmitted is passed through LPF. The output of the LPF’s are connected to the rotating sampling switch or commutator. It takes the sample from each signal per resolution and rotates at the rate of fs. At the receiver, the commutator separates the time multiplexed input channels. These channel signals are then passed through low pass reconstruction filter.

SAMPLING FREQUENCY:

If the highest signal frequency present in all channels is ‘w’, then by sampling theorem, the sampling frequency fs should be,

fs ≥2w ---------> (1)

FRAME:

If the sampling frequency is fs, then the time space between successive samples from any one input will be,

Ts = 1/fS ---------> (2)

Ts ≤ 1/2w ---------> (3)

Thus the time interval Ts contains one sample from each input. This time interval is called frame.

GUARD TIME:

The baseband filtering gives rise to inter channel crosstalk from one sample value to next. This interference can be reduced by increasing the distance between individual signal samples. The minimum distance between the individual signal samples to avoid crosstalk is called guard time.

Slow speed TDM, as often used in radio telemetry is produced simply with rotating mechanical switches. A number of channels are fed simultaneously to the switch in the transmitter while the output is taken over from the rotating rotor.

This rotor rotates slowly and remains in contact with each other channel for a predetermined period during which, the output of that channel is the only one passed on for transmission. There is a corresponding rotating switch in the receiver, synchronized to the one in transmitter, which reverses the process to receive and the 24 channels are duly separated.

In the receiver, the output of the main detector is fed simultaneously to 24 AND gates. An AND gate is a simple device having one output and 2 or more input terminals, so arranged that an output is obtained only if all input signals are present. In this case, each gate has two input terminals and the second input to each gate is provided from a synchronized clock gating generator, which is a monostable multivibrator producing rectangular pulses of 5µs duration, 8000 times per second. Delay lines or circuits are used once again, with the gating pulse to first gate is not delayed at all, that to second gate delayed by 5 µs and so on. In this fashion, each gate is open only during the appropriate time intervals and the 24 channels are duly separated.

If the transmission is by wire, then 1.544 Mbps pulse train is the signal sent, but if cable or radio communication is used, the pulse train either modulate the carrier or else it is further multiplexed with similar pulse trains, all combined together into a higher TDM hierarchical layer. The circuit diagram in figure 4.1 shows the experimental setup of Time division multiplexing. The signals to be multiplexed are applied at the input of the buffer. At the output of the buffer the signals are sampled by the samples from the decoder. The signals are then sampled and applied to the low pass filters. The demultiplexed and demodulated signals are obtained at the output of low pass filters.

LAYOUT DIAGRAM OF TRAINER KIT:

Figure 4.1 Experimental Setup of Time Division Multiplexing Trainer Kit

PROCEDURE:

1. Connect the four channel inputs 250Hz, 500Hz, 1KHz, 2KHz to the input of the transmitter CH0, CH1, CH2 and CH3 respectively.

2. Connect TXCLOCK(Transmitter Clock) to RXCLOCK(Receiver Clock).

3. Connect TXCH0(Transmitter Sync) to RXCh0(Receiver Sync).

4. Connect the TXD(Transmitter data) to RXD( Receiver data).

5. Observe the Multiplexed data at TDX, Transmitter Clock at TXCLOCK and transmitter sync at TXCH0.

6. Observe the demultiplexed signals at the receiver across the output of the fourth order low pass filter at the CH0, CH1, CH2 and CH3 respectively.

TABULATION:

SL.NO SIGNALS AMPLITUDE

(Vp-p)

FREQUENCY

(Hz)

Bit Rate or Sample Rate=

RESULT:

MODEL WAVEFORMS:

MULTIPLEXED AND DEMULTIPLEXED SIGNALS:

TDM OF SAMPLES OF SIGNAL:

1 Explain the term Multiplexing applied in Communication Engineering.

Multiplexing refers to combining various analog or digital signals into one and sending them over a shared medium. It is aimed at sharing resources (like bandwidth)

2. List the various types of Multiplexing.

1. Space division multiplexing 2. Frequency division multiplexing 3. Time division multiplexing4. Polarization-division multiplexing5. Orbital angular momentum multiplexing6. Code-division multiplexing

3. What is known as Guard time/interval?In telecommunications, guard intervals are used to ensure that distinct transmissions do not interfere with one another. These transmissions may belong to different users (as in TDMA) or to the same user (as in OFDM).

The purpose of the guard interval is to introduce immunity to propagation delays, echoes and reflections, to which digital data is normally very sensitive.

4. What is the basic difference between synchronous and asynchronous TDM?Synchronous TDM is the standard operation mode of TDM operation where each multiplexed channel has a specific size slot allowed for it, whether data exist to transmit or not. This is very wasteful of bandwidth space if you have channels that only occasionally transmit. Asynchronous TDM is actually called Statistical TDM. It does not reserve a time slot for each channel, rather it assigns a slot when the channel is requiring data to be sent or received. This allows for variable bandwidth per channel as required.

5. What is the minimum bandwidth required for TDM of PAM signal.The minimum bandwidth required is equal to the maximum frequency among the signals to be multiplexed.

EXP.NO:

DATE:

AMPLITUDE SHIFT KEYING

AIM:

To construct the amplitude shift keying modulator & demodulator circuit and study their operation.

DEVICES, COMPONENTS AND EQUIPMENT REQUIRED:

S.NO NAME TYPE/RANGE QUANTITY

1.

2.

3.

4.

5.

6.

7.

8.

Operational amplifier

Field effect transistor

Diode

Resistors

Capacitors

Signal generator

Regulated power supply

Cathode ray oscilloscope

IC 741

BFW10

0A79

112KΩ,82KΩ,

47KΩ,4.7KΩ

10KΩ,1KΩ

10nF,1nF

(0-10M)Hz

+15V,(0-30)V

1MHz

2

1

1

1

1

5,2

1,2

1

1

1

SPECIFICATIONS OF IC741, BFW10, 0A79:

Refer the enclosed datasheet in appendix.

THEORY:

AMPLITUDE SHIFT KEYING:

Amplitude shift keying can be defined as the process whereby the amplitude of an RF carrier is varied in accordance with the information to be transmitted. The general form of the carrier wave, C(t), is as follows:

C(t) = A(t) cosθ(t)

Where A(t) is the time-varying amplitude and θ(t) is the time-varying angle. It is convenient to write

θ(t) = ω0t + Φ(t)

so that C(t) = A(t) cos[ω0t +Φ(t)] ---------- eqn (5.1)

where C(t) = Carrier sine wave

A(t) = Time varying amplitude

ω0t = Time varying angle

Φ(t) = Phase difference

In amplitude shift keying modulator shown in the Figure 5.2, the n-channel JFET is forward biased when the message signal is positive, i.e. a logical ‘1’.The barrier potential of the JFET decreases and there is current flow from the drain to source. Thus, FET offers zero impedance or it acts like a closed switch. The carrier signal is grounded through the short circuit. Hence the output is low. When the input signal is negative, i.e. at logical ‘0’, the JFET is reverse biased, thereby preventing current flow from drain to source. The JFET now acts like an open switch. So, the carrier signal is conducted to the input stage of the OP-AMP inverting amplifier which produces a phase shift of 180o. Thus, the amplitude of the carrier is keyed in accordance with the amplitude of the message signal.

ASK DEMODULATION:

Envelope detectors built around rectifiers are normally used for ASK detection. The Figure 5.1 illustrates an ASK detector employing a diode detector.

Figure 5.1 ASK detector

The diode acts as a rectifier and can be considered to be an “ON” switch when the input voltage is positive, allowing the capacitor C to charge up to the peak of the carrier input. During the negative half of the input, the diode is “OFF”, but the capacitor holds the positive charge previously received, so the output voltage remains at the peak positive value of carrier wave.

There will, in fact be some discharge of C, producing an RF ripple on the output waveform, which must filtered out as the input voltage rises, the capacitor voltage has no difficulty in following this, but during the downward swing, the capacitor may not discharge fast enough, unless a discharge path is provided by a resistor. The resistor R does this job.

The time constant of the RC load has to be short enough to allow the output voltage to follow the modulation cycle, and yet long enough to maintain a relatively high output voltage. The threshold detector compares the envelope (which will contain some ripple), with the reference of 5V and switches to either “HIGH” or “LOW” state, thereby recovering the original data from the ASK wave. The ASK demodulator circuit is shown in Figure 5.3. The model waveforms of ASK modulator and demodulator are given in Figure 5.4 and Figure 5.5 respectively.

ADVANTAGE:

ASK is the simplest kind of modulation to generate and detect.

DISADVANTAGES:

ASK is very susceptible to noise interference - noise usually (only) affects the amplitude, therefore ASK is the modulation technique most affected by noise

Low power efficiency

CIRCUIT DIAGRAM:

Figure 5.2: ASK Modulator circuit

Figure 5.3: ASK Demodulator circuit

TABULATION:

SL.NO SIGNALS AMPLITUDE

(Vp-p)

FREQUENCY

(Hz)

Bit Rate =

PROCEDURE:

(1) The ASK modulator and demodulator circuits are constructed as per the circuit diagrams shown in Figures 5.2 and 5.3 respectively.

(2) The data signal is applied to the ASK modulator.

(3) The data, carrier and the modulated signals are studied and the values are tabulated.

(4) The modulated output is fed as input to the demodulator circuit shown in Figure 5.3.

(5) The demodulated signal is studied, tabulated and drawn.

RESULT:

SIMULATION WAVEFORMS:

Figure 5.4: INFORMATION, CARRIER, ASK MODULATED

SIGNALS

Figure 5.5: ASK MODULATED AND DEMODULATED SIGNALS

1. What is the relation between bits and symbols?

A symbol is a group of bits. The bit stream to be transmitted is serial to parallel multiplexed onto a stream of symbols with q bits per symbol (discrete 2q levels).

2. What is m-ary modulation or m-ary signalling?

M-ary signalling is a type of digital modulation where instead of tramsmitting one bit at a time, two or more bits are transmitted simultaneously. This type of transmission results in reduced channel bandwidth.

3.Why is ASK called as on-off keying?

Since in ASK modulation the digital data is represented as the presence and absence of the carrier wave, this type of modulation is called on-off modulation.

4.What is the basic difference between baseband modulation and bandpass modulation?

The baseband is defined as a transmission signal that contains more than just a single frequency from 0 Hz to the highest frequency component.The passband is the output of a band-pass filter. It is a signal corresponds to the settings of the band-pass filter. While baseband is the original signal, passband is the filtered signal. Passband is more technically defined as the portion of the spectrum between limiting frequencies with minimum relative loss or maximum relative gain.

5.What are the advantages and disadvantages of ASK modulation?

ADVANTAGE:

Reduction in the amount of energy required to transmit information.

DISADVANTAGE:

Easily influenced to noise interference since information is carried in amplitude.

EXP.NO:

DATE:

FREQUENCY SHIFT KEYING MODULATOR AND DEMODULATOR

AIM:

To design and construct a frequency shift keying modulator & de modulator circuit. Study its operations and obtain their output waveforms

DEVICES, COMPONENTS AND EQUIPMENT REQUIRED:

S.NO NAME TYPE/RANGE QUANTITY

1

2

3

4

5

6

7

Integrated circuits

Resistors

Capacitors

Potentiometer

Function generator

Regulated power supply

Cathode ray oscilloscope

XR 2206

NE 565

5kΩ, 934kΩ,

785kΩ, 27kΩ

5.1kΩ, 680Ω

10kΩ

1nf, 0.05nf

0.01uf, 1uf

200Ω

(0-1M)Hz

(0-30)V

30MHz

1

1

1

1

2

5

1

2

1

1

1

1

DESIGN:

Mark frequency= 1 / (R1*C) ------------ (6.1)

Space frequency= 1/(R2*C) ------------ (6.2)

THEORY:

Frequency shift keying is a relatively simple low performance form of digital communication. It is a constant envelope from of angle modulation similar to conventional FM except that the modulating signal is a digital signal.

FSK GENERATION USING XR2206:

FSK is widely used in data interface or acoustical coupler type of modem system. In monolithic IC oscillators FSK capability is obtained by using a current controlled oscillator and keying its control current between one or more programmed levels which are set by the external registers. This results in output waveform which is continuous during the transition between mark and space frequencies. XR2206 produces two discrete frequencies f1, f2 and has one bit keying logic unit.

TIMING RESISTOR MODE:

The circuit connection of XR 2206 in timing resistor is shown. The data input is applied to pin 9. A high level signal selects the frequency and low level signal selects the frequency for optimum capability R5 and R6 should b taken such that their value lie in the range of 10kohm to 100kohm. In appreciation of EX OR 2206 the applications were minimal distortion is not necessary, pin 15 and 16 maybe left open circuitedFrequency shift keying(FSK) is a form of orthogonal multi plus modulation were the pulses are not limited by band.

ADVANTAGES:

The primary advantages of FSK are in areas other than spectral efficiency

INCOHERENCE:

Carrier recovery for PAM signals for some trivial, especially on certain channels where the received carrier phase varies rapidly. Effective FSK receivers can be designed that make no attempt to recover the carrier phase. Such receivers are said to be incoherent.

EASE OF IMPLEMENTATION:

Very simple PSK modems are possible mainly because of incoherence detection is feasible. In addition, it is also sometimes possible to avoid timing recovery.

IMMUNITY FROM CERTAIN NON LINEARITIES:

Most FSK modulation techniques result in a constant envelope, in which information is carried by the zero crossings of the signal alone.

WORKING:

FSK MODULATOR:

The digital input is given at pin 9 of EX OR 2206. The timing resistor which decides mark and space frequencies are connected to pins 8 and 7 respectively.Based on the digital inputs the currents switches select the timing resistor. If digital ‘1’s is the input Rin resistor is selected and if digital ‘0’ is the input resistor Rs is selected. Based on the resistor selected and different current flows for ‘0’ and ‘1’ and thereby the VCO produces different digital inputs ‘0’ and ‘1’. The output of the VCO is given to the multiplier and sign shaper and then when amplifier whose output is the required sine/triangular wave output.

FSK DEMODULATOR:

The demodulator is NE 565 which is a PLL. It consists of phase comparator, error amplifier, low pass filter and voltage controlled oscillator. The phase comparator compares the incoming frequency with that of the reference generated by the VCO. It is called the free running frequency (carrier frequency). If there is a change in frequency accordingly the error is fed to the VCO and subsequently the VCO frequency tries to lock the incoming frequency.The output at pin 7 is the demodulated output and it is given to pin 3 and reference output is given to pin 2 of the comparator which gives the base band signal back of the demodulation. FSK modulations are employed in modems.

PROCEDURE:

1. The circuit connections are given as per the circuit diagram in figure 6.1.

2. The carrier signal output is observed at pin 2 without giving the data signal and by grounding pin8 and connecting 10 kΩ to pin 7.

3. The data signal is set at desired amplitude and frequency by giving appropriate value of resistors at pin 7 and 8.

4. The FSK Modulated signal is observed at pin 2, noted and recorded.

5. Logic 1 and logic 0 DC voltages are applied at pin 9 to note down the mark and space frequency.

6. The FSK modulated output which is obtained is given to pin 2 as per the circuit diagram shown in figure 6.2

7. The demodulated signal is observed at pin 6 of Op-Amp, noted and recorded.

TABULATION

SL.NO SIGNALS AMPLITUDE (Vp-p) FREQUENCY (Hz)

Bit rate=

RESULT:

CIRCUIT DIAGRAM:

Fig 6.1: FSK Modulation circuit

Fig 6.2: FSK Demodulation Circuit

MODEL GRAPH:

Figure: 6.3 waveforms of FSK Modulator and Demodulator

1. What is the condition for two signals to be orthogonal?

The condition for 2 signals to be orthogonal if their basis function s1(t) and s2(t) satisfy the condition

∫0

T

s j ( t )∗sk (t ) dt=K j0≤ t ≤ T , j , k=1 , ……

When the Kj constants are non-zero, the signal space is called orthogonal.

2. Compare the probability of bit error of coherently detected bfsk and that of non coherently detected bfsk

Coherently detected BFSK :

PB = 1

√ 2 π∫

√E b/No

α

exp (−u2

2¿)du¿

= Q(√ EbNo

)

Non coherently detected BFSK :

PB = 12

exp(−12

EbNo )

3. What is continuous phase fsk?

Continuous-phase frequency-shift keying (CPFSK) is a commonly used variation of frequency-shift keying (FSK), which is itself a special case of Analog frequency modulation. FSK is a method of modulating digital data onto a sinusoidal carrier wave, encoding the information present in the data to variations in the carrier's instantaneous frequency between one of two frequencies (referred to as the space frequency and mark frequency). In general, a standard FSK signal does not have continuous phase, as the modulated waveform switches instantaneously between two sinusoids with different frequencies.

4. What is the basic difference between cpfsk and msk?

MSK is the special form of binary CPFSK with modulation index h= ½.

EXP.NO:

DATE:

FREQUENCY DIVISION MULTIPLEXING

AIM:

To generate Frequency Division Multiplexed signals and to obtain their De-multiplexed output.

DEVICES, COMPONENTS AND EQUIPMENT REQUIRED:

DESIGN

FORMULA:

DESIGN:

SL.NO NAME TYPE/RANGE QUANTITY

1

2

3

4

5

6

Integrated circuits

Resistors

Capacitors

Function Generator

Regulated Power Supply

Cathode Ray Oscilloscope

IC741

As per the design

(0-1M)Hz

(0-15) V

(0-30M)Hz

03

As per the design

02

02

01

THEORY:

Frequency division multiplexing is the position of signal spectra in frequency such that each signal spectrum

can be separated out from all the others by filtering. FDM does not preclude the use of other modulating

methods. There are N signals in frequency, each is band limited to fm Hz. In order to separate N signals in

frequency, each is modulated with a carrier frequency fc1, fc2… fcm. Using DSB-LC, the spectral density of

every modulated signal has a bandwidth of 2fm and each is centered at various carrier frequencies fc1, fc2,…,

fcn as shown in Fig8.1. These carrier frequencies are chosen far enough apart such that each signal spectral

density is separated from all the others. In Telephony, the most widely used method of modulation in FDM

is single sideband modulation, which, in the case of voice signals, requires a bandwidth that is

approximately equal to that of the original voice signal. Each voice input is usually assigned a bandwidth of

4 KHz. The band pass filters following the modulators are used to restrict the band of each modulated signal

to its prescribed range. The resulting band pass filter outputs are combined in parallel to form the input to

the common channel.At the receiving terminal, a bank of band pass filters, with their inputs connected in

parallel, is used to separate the message signals on a frequency-occupancy basis. The original message

signals are recovered by individual demodulators as shown in Fig8.2

Figure 7.1 Block diagram of FDM Transmitter

Figure 7.2 Block diagram of FDM Receiver

CIRCUIT DIAGRAM:

Figure 7.3 Experimental setup of Frequency Division Multiplexer and De multiplexer

TABULATION:

SL.No. SIGNAL AMPLITUDE(Vp-p)

FREQUENCY (Hz)

TIME PERIOD (ms)

MODEL WAVEFORMS:

INPUT SIGNAL Ι:

INPUT SIGNAL ΙΙ:

FRREQUENCY DIVISION MULTIPLEXER OUTPUT:

FRREQUENCY DIVISION DEMULTIPLEXER OUTPUT:

DEMULTIPLEXED OUTPUT I:

DEMULTIPLEXED OUTPUT ΙΙ:

WORKING:

The two signals which are to be multiplexed are given as inputs to the op-amp adder circuit which acts as

frequency division multiplexer. The frequency division multiplexed output is given as input simultaneously

to the low pass high pass filters as shown in Fig8.3. Thus the output signals obtained are similar to the input

signals which are multiplexed.

PROCEDURE:

1. Connections are given as per the circuit diagram.

2. The 2 signals to be multiplexed are fed to op-amp which performs adder operation which acts as a multiplexer.

3. The frequency division multiplexed signal is applied simultaneous to the LPF & HPF.

4. The filters are designed in such a way that low frequency signal is passed only through the LPF and high frequency signal is passed only through HPF.

5. The outputs obtained are similar to the multiplexed signals.

RESULT:

1. DIFFERENTIATE BETWEEN FDM AND TDM.

The primary difference between FDM and TDM is how they divide the channel. FDM divides the channel into two or more frequency ranges that do not overlap, while TDM divides and allocates certain time periods to each channel in an alternating manner. Due to this fact, we can say that for TDM, each signal uses all of the bandwidth some of the time, while for FDM, each signal uses a small portion of the bandwidth all of the time. TDM provides greater flexibility and efficiency whereas FDM lacks flexibility.

2. GIVE SOME EXAMPLES WHERE FDM IS APPLIED

Commercial broadcast radio (AM and FM radio) FDM is used mainly for analog transmissions. It can be used over both wired and wireless

mediums. Another application that uses FDM is cable TV

3. WHAT IS GUARD BAND?

In FDM, a number of signals are sent simultaneously on the same medium by allocating separate frequency band or channel to each signal. The spectral region between these successive channels is called guard band which are used to avoid interference between the two channels