Beruflich Dokumente
Kultur Dokumente
NO:
DATE
TYPE/RANGE
QUANTITY
30MHz
Patch cords
(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--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--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--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
RETURN TO ZERO-AMI:
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 bandwidth
2. Power efficiency
3. Error detection and correction capability
4. Favorable power spectral density
5. Adequate timing content
6. 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 variable
The time-bandwidth product is thus a robust measure of combined time and frequency spread
of 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:
NAME
TYPE/RANGE
QUANTITY
1.
DCLT-003
2.
DCLT-004
3.
Patch cords
10(approx.)
4.
(0-30)MHz
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,
BLOCK DIAGRAMS:
LAYOUT DIAGRAM:
Figure 2.3 Pulse Code Modulator and Demodulator trainer kit layout diagram
TABULATION:
SL.NO
SIGNALS
Input Signal-A
Input Signal-B
Demodulated Signal A
Demodulated Signal B
Sampling rate =
AMPLITUDE (Vp-p)
FREQUENCY (Hz)
( samples/second )
RESULT:
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.
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.
EXP.NO:
DATE:
DELTA MODULATION
AIM:
To construct a Delta Modulator and to study its performance.
TYPE/RANGE QUANTITY
Op-amp
IC 741
Dual D-Flipflop
IC 4013
Resistors
10K
4.7 K
100 K
Capacitors
0.01F
(0-30)V
Function Generator
(0-1M)Hz
30MHz
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.
=2
(3.1)
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(
FREQUENCY (Hz)
EXP.NO:
DATE:
SL.NO
NAME
TYPE/RANGE
QUANTITY
DCLT-002
Power supply
(0-30)V
Patch Cords
(10-15)
Input channels : 4
Switching
: TDM
Modulation
: PAM
Sampling rate
: 32 KHz
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 LPFs 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 5s 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 s 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.
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 FREQUENCY
(Vp-p)
RESULT:
(Hz)
MODEL WAVEFORMS:
MULTIPLEXED AND DEMULTIPLEXED SIGNALS:
EXP.NO:
DATE:
NAME
TYPE/RANGE
QUANTITY
1.
Operational amplifier
IC 741
2.
BFW10
3.
Diode
0A79
4.
Resistors
112K,82K,
47K,4.7K
10K,1K
5,2
5.
Capacitors
10nF,1nF
1,2
6.
Signal generator
(0-10M)Hz
7.
+15V,(0-30)V
8.
1MHz
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) cos(t)
Where A(t) is the time-varying amplitude and (t) is the time-varying angle. It is convenient to write
(t) = 0t + (t)
so that
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.
CIRCUIT DIAGRAM:
TABULATION:
SL.NO
SIGNALS
AMPLITUDE
FREQUENCY
(Vp-p)
(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:
SIGNALS
EXP.NO:
DATE:
NAME
TYPE/RANGE
QUANTITY
Integrated circuits
XR 2206
NE 565
5k, 934k,
785k, 27k
5.1k, 680
10k
1nf, 0.05nf
0.01uf, 1uf
Resistors
Capacitors
Potentiometer
200
Function generator
(0-1M)Hz
Regulated
supply
Cathode
oscilloscope
power (0-30)V
ray 30MHz
DESIGN:
Mark frequency= 1 / (R1*C)
------------ (6.1)
------------ (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.
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 k 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
Bit rate=
RESULT:
CIRCUIT DIAGRAM:
MODEL GRAPH:
The condition for 2 signals to be orthogonal if their basis function s1(t) and s2(t) satisfy the condition
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
EXP.NO:
DATE:
NAME
Integrated circuits
Resistors
TYPE/RANGE
IC741
Capacitors
Function Generator
QUANTITY
03
DESIGN
(0-1M)Hz
02
(0-15) V
02
FORMUL
A:
DESIGN:
(0-30M)Hz
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
CIRCUIT DIAGRAM:
TABULATION:
SL.No. SIGNAL
AMPLITUDE FREQUENCY
(Vp-p)
(Hz)
TIME PERIOD
(ms)
MODEL WAVEFORMS:
INPUT SIGNAL :
INPUT SIGNAL :
DEMULTIPLEXED OUTPUT :
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:
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
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