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TEXT BOOK:
1. Digital communications, Simon Haykin, John Wiley India Pvt. Ltd, 2008.
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REFERENCE BOOKS:
1. Digital and Analog communication systems, Simon Haykin, John Wildy India Lts, 2008
2. An introduction to Analog and Digital Communication, K. Sam Shanmugam, John Wiley India
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Pvt. Ltd, 2008.
3. Digital communications - Bernard Sklar: Pearson education 2007
PREPARED BY:
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RAGHUDATHESH G P
Asst Prof
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Cell: +917411459249
Mail: datheshraghubooks@gmail.com
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Quotes:
Introduction:
Analog waveforms or signals are sampled into pulses. These analog pulses are also called
carrier pulses. When amplitude of these pulses varies according to amplitude of analog
waveform, it becomes pulse amplitude modulation (PAM).
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The pulses of PAM can be converted to digital form. Thus the analog waveform is
converted to sequence of binary digital format. It is also called pulse code modulation.
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When the analog pulse is converted to digital form, it is quantized to nearest digital level.
This introduces quantization noise in the signal.
The digital waveform coding techniques are always selected on the basis of quantization
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noise and bandwidth of transmission.
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Definition: In pulse-amplitude modulation (PAM), the amplitude of a carrier consisting
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of a periodic train of rectangular pulses is varied in proportion to sample values of a
message signal. In this type modulation, the pulse duration is held constant.
The carrier is a train of rectangular pulses. By making the amplitude of each rectangular
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pulse the same as the value of the message signal at the leading edge of the pulse, PAM
so defined is exactly the same as flat-top sampling. Thus PAM Wave s(t), is defined as
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Here
g(nTs) = Sample values of the message (modulating) signal g(t).
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Disadvantages of PAM:
1. As we have seen just now, the bandwidth needed for transmission of PAM signal is very
large compared to its maximum frequency content.
2. The amplitude of PAM pulses varies according to modulating signal Therefore
interference of noise is maximum for the PAM signal and this noise cannot be removed
very easily.
3. Since amplitude of PAM signal varies, this also varies the peak power required by the
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transmitter with modulating signal.
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Other Forms of Pulse Modulation:
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There are two more types of pulse modulation other than PAM:
1. Pulse Duration Modulation (PDM):
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In this technique the width of the pulse changes according to amplitude of the
modulating signal at sampling instant.
2. Pulse position Modulation(PPM):
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In this technique the position of the pulse changes according to amplitude of the
modulating signal at sampling instant.
Pulse position modulation (PPM) and pulse duration modulation (PDM or PWM) both
modulate the time parameter of the pulses. PPM has fixed width pulses where as width of
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Amplitude of the pulse is Width of the pulse is The relative position of pulse
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2 proportional to amplitude of proportional to amplitude of is proportional to amplitude
modulating signal. modulating signal. of modulating signal.
The bandwidth of the Bandwidth of transmission Bandwidth of transmission
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3 transmission channel depends on channel depends on rise time channel depends on rising
width of the pulse. of the pulse time of the pulse.
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The instantaneous power of the The instantaneous power of The instantaneous power of
4 transmitter varies. the transmitter varies. the transmitter remains
constant.
Noise interference is high. Noise interference is Noise interference is
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minimum. minimum.
6 System is complex. Simple to implement. Simple to implement.
Similar to amplitude modulation Similar to frequency Similar to phase modulation.
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modulation.
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communication channel for only a fraction of the sampling interval on a periodic basis.
Due to above reason, some of the time interval between adjacent pulses of the PAM wave
is cleared for use by other independent message signals on a time-shared basis.
By so doing, we obtain a time-division multiplex system (TDM), which enables the joint
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Each input message signal is first restricted in bandwidth by a low-pass pre-alias filter to
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remove the frequencies that are nonessential to an adequate signal representation.
The pre-alias filter outputs are then applied to a commutator, which is usually
implemented using electronic switching circuitry.
The function of the commutator is two-fold:
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1. To take a narrow sample of each of the N input messages at a rate fs that is
slightly higher than 2W, where W is the cutoff frequency of the pre-alias filter.
2. To sequentially interleave these N samples inside a sampling interval T s = 1/fs.
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Then the sampling rate for each message signal is determined accordance with the
sampling theorem.
Let
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At the receiving end of the system, the received signal is applied to a pulse amplitude
demodulator, which performs the reverse operation of the pulse amplitude modulator.
The short pulses produced at the pulse demodulator output are distributed to the
appropriate low-pass reconstruction filters by means of a decommutator, which operates
in synchronism with the commutator in the transmitter.
The synchronization is essential for a satisfactory operation of the TDM system, and
provisions have to be made for it.
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The figure below illustrating TDM for 2 message signals.
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Drawbacks of TDM:
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4. Signaling rate = 2 × Transmission Bandwidth.
5. Speed of the commutator in revolution per second(rps) = 2W
Here W = minimum bandwidth of the message signal
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6. Speed of commutator (samples/sec) = Total number of segments × Speed of
commutator(rps)
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7. Minimum transmission bandwidth = ½ [sum of nyquist rate]
8. Angle of separation b/w each segment(pole)= 360/Total number of segments(n).
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Problems on TDM
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1. A Signal m1(t) is band limited to 3.6khz and three signals m2(t), m3(t) and m4(t) are band
limited to 1.2 kHz each. These signals are to be transmitted by means of TDM. Sketch set
up a scheme for realizing this multiplexing requirement with each signal sampled at its
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Solution
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m3(t) m1(t)
2.4 kHz
m1(t) PAM Channel
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m2(t)
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m4(t) m1(t)
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Thus if commutator is rotated at 2400 revolution/second then in each revolution, we get
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one sample each for m2(t), m3(t) and m4(t) and three sample for m1(t).
= 7.2 kHz.
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2. Eight message signals are sampled and time multiplexed using PAM. The time
multiplexed signal is passed through a LPF before transmission. Six of the input signals
have a bandwidth of 4 kHz and the other two are band-limited to 12 kHz.
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(a) What is the minimum overall sampling rate if all the messages are sampled at the same
rate?
(b) Design an asynchronous TDM for this application.
(c) Compare the transmission bandwidth requirements of parts (a) and (b).
Solution:
(a) If all the messages are sampled at the same rate, then f, = 2 x 12,000 = 24,000
samples/second per message.
Hence, overall sampling rate = 8 x 24, 000 = 192, 000 samples/second.
(b) Let g1(t), g2(t) be the 12 kHz message signals and g3(t), g4(t), ….,g8(t) be the 4 kHz message
signals. Also let the switch make 8000 rotations/second. It samples 4 kHz messages once per
rotation and the 12 kHz messages three times per rotation.
g1(t)
g3(t)
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g8(t) g2(t)
g4(t)
g2(t) Commutator
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PAM
g7(t) Channel
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g1(t)
g6(t) g1(t)
g2(t) g5(t)
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(c) Synchronous sampling rate = 192,000 samples/second.
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BW = ½[Synchronous sampling rate] = ½[192,000] = 96 kHz
Asynchronous sampling rate = 8000 x 12 = 96,000 samples/second.
BW = ½[Synchronous sampling rate] = ½[96,000] = 48 kHz.
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3. A signal x1(t) is bandlimited to 3 kHz. There are three more signals x2(t), x3(t) and x4(t)
which are bandlimited to 1 kHz each. These signals are to be transmitted by a TDM
system. (i) Design a TDM scheme where each signal is sampled at its Nyquist rate. (ii) What
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must be the speed of the commutator? (iii) Calculate the minimum transmission bandwidth
of the channel.
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Solution:
(i) Table below shows different message signals with corresponding Nyquist rates.
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If the sampling commutator rotates at the rate of 2000 rotations per second then the signals x2(t),
x3(t) and x4(t) will be sampled at their Nyquist rate. But, we have to sample x1(t) also at its
Nyquist rate which is three times higher than that of the other three.
In order to achieve this, we should sample x1(t) three times in one rotation of the commutator.
Therefore, the commutator must have atleast 6 poles connected to the signals as shown in figure
x3(t)
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x1(t)
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PAM
Channel
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x4(t)
x1(t) x2(t)
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(ii) The speed of rotation of the commutator is 2000 rotations/sec.
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(iii) Number of samples produced per second is calculated as under:
X1(t) produces 3 x 2000 = 6000 samples/sec.
x2(t), x3(t) and x4(t) produce 2000 samples/sec. each.
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4. Six message signals each of bandwidth 5 kHz are time division multiplexed and
transmitted. Determine the signaling rate and the minimum channel bandwidth of the
PAM/TDM channel.
Solution:
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5. Twenty-four voice signals are sampled uniformly and then time division multiplexed.
The sampling operation uses flat top samples with 1 µs duration. The multiplexing
operation includes provision for synchronization by adding an extra pulse of appropriate
amplitude and 1 µs duration. The highest frequency component of each voice signal is 3.4
kHz.
(i) Assuming a sampling rate of 8 kHz, calculate the spacing between successive pulses of
the multiplexed signal.
(ii) Repeat (i) assuming the use of Nyquist rate sampling.
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Solution:
(i) Given that Sampling rate = 8 kHz = 8000 samples/sec.
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There are 24 voice signals + 1 synchronizing pulse.
Pulse width of each voice channel and synchronizing pulse is 1 µs.
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Now, time taken by the commutator for 1 rotation
Number of pulses produced in 1 rotation = 24 + 1 = 25
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Therefore, the leading edges of the pulses are at distance as shown in figure
below
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This means that 6800 samples are produced per second. One rotation of commutator takes
1/6800 = 147 µs time.
Therefore, 147 µ sec corresponds to 25 pulses.
Therefore, 1 pulse corresponds to 5.88 t sec.
As the pulse width of each pulse is 1 µ sec, the spacing between adjacent pulses will be 4.88
µsec and if we assume τ= 0 then the spacing between the adjacent pulses will be 5.88 µsec.
Pulse-code modulation (PCM) provides one method to generate a coded version of the
signal.
Broadly speaking PCM is essentially analog-to-digital conversion where the analog
samples are represented by digital words in a serial bit stream.
The use of digital representation of analog signals (e.g., voice, video) offers us the
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following advantages:
1. Ruggedness to transmission noise and interference.
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2. Relatively inexpensive digital circuitry may be used extensively in the system.
3. Digital information can be encrypted for security, coded against errors and
compressed to reduce storage and transmission costs.
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4. The possibility of a uniform format for different kinds of pulse signals.
5. In long-distance digital telephone systems, a clean PCM waveform can be generated
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using regenerative repeaters kept at regular locations all along the communication
route.
6. PCM signals derived from all types of analog sources (audio, video, etc.) may be
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interleaved with data signals (e.g., from digital computers) and transmitted over a
common channel. This technique is called time-division multiplexing.
7. The noise performance of a digital communication system can be superior to that of
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Pulse-code modulation systems are complex in that the message signal is subjected to a
large number of operations.
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The essential operations in the transmitter of a PCM system are sampling, quantizing,
and encoding, as shown in the figure below.
The sampling, quantizing, and encoding operations are, usually performed in the same
circuit, which is called an analog-to-digital converter.
Regeneration of impaired signals occurs at intermediate points along the transmission
path (channel) as indicated in the Figure below.
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At the receiver, the essential operations consist of one last stage of regeneration
followed by decoding, then demodulation of the train of quantized samples, as in the
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Figure below.
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The operations of decoding and reconstruction are usually performed in the same circuit,
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Note:
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The only part of pulse-code modulation that conforms to this definition sampling.
The subsequent use of quantization, which is basic to pulse-code modulation, introduces
a signal distortion that has no counterpart in conventional modulation.
1. Sampling:
The incoming message wave is sampled with a train of narrow rectangular pulses so as to
closely approximate the instantaneous sampling process.
To ensure perfect reconstruction of the message at the receiver, the sampling rate must be
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greater than twice the highest frequency component W of the message wave (in
accordance with the sampling theorem).
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Low-pass pre-alias filter is used at the front end of the sampler in order exclude
frequencies greater than W before sampling.
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2. Quantizing:
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An analog signal, such as voice, has a continuous range of amplitudes and therefore its
samples cover a continuous amplitude range.
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It is not necessary in fact to transmit the exact amplitudes of the samples. Any human
sense (the ear or the eye), as ultimate receiver, can detect only finite intensity differences.
The existence of a finite number of discrete amplitude levels is a basic condition of PCM.
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Definition: The conversion of an analog (continuous) sample of the signal into a digital
(discrete) form is called the quantizing process.
Graphically, the quantizing process means that a straight line representing the relation
between the input and the output of a linear analog system is replaced by a transfer
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staircase, and
2. The output is assigned a discrete value selected from a finite set of
representation levels or reconstruction values that are aligned with the "treads"
of the staircase.
For a uniform quantizer, whose characterized as in Figure below, the separation between
the decision thresholds and the separation between the representation levels of the
quantizer have a common value called the step size.
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According to the staircase-like transfer characteristic of Figure above, the decision
thresholds of the quantizer are located at , , , . . . , and the
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representation levels are located at 0, , , . . . , where is the step size.
A uniform quantizer characterized in this way is referred to as a symmetric
quantizer of the midtread type, because the origin lies in the middle of a tread of
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the staircase.
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quantizer of the midriser type, because in this case the origin lies in the middle of a
riser of the staircase.
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A quantizer of the midtread or midriser type, as defined, is memoryless in the quantizer
output is determined only by the value of a corresponding sample, independently of
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earlier (or later) analog samples applied to the input.
The memoryless quantizer is the simplest and most often used quantizer.
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The transfer characteristics of Symmetric quantizer of the midtread type and Symmetric
quantizer of the midriser type, we have included a parameter labeled the overload level,
the absolute value of which is one half of peak-to-peak range of input sample values.
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The number of intervals into which the peak-to-peak excursion is divided, or equivalently
the number of representation levels, is equal to twice the absolute value of the overload
level divided by the step size.
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Thus, for an analog input sample that lies anywhere inside an interval of either transfer
characteristic, the quantizer produces a discrete output equal to the midvalue of the pair
of decision thresh in question.
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In so doing, however, a quantization error is introduced, the value of which equals the
difference between the output and input values of the quantizer.
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Both figures show the variations of the quantization error with the input for the two
uniform quantizer types. In both cases, we see that the maximum instantaneous value of
this error is half of one step size, total range of variation is from minus half a step to plus
half a step.
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3. Encoding:
Binary code:
Each symbol may be either of two distinct values or kinds, such as the
presence or absence of a pulse.
The two symbols of a binary code customarily denoted as 0 and 1.
Ternary code:
Each symbol may be one of three distinct values or kinds.
So on for other codes.
Generally binary codes are used due to maximum advantage over the effects of noise in a
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transmission medium is obtained by using a binary code, because a binary symbol
withstands a relatively high level of noise and is easy to regenerate.
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In a binary code, each code-word consists of n bits. Then, such a code, we may represent
a total of 2n distinct numbers.
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Ex., a sample quantized into one of 24 = 16 levels may be represented by a 4-bit -word.
There are several formats (waveforms) for the representation of binary sequences
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produced by analog-to-digital conversion (or by other sources). Figure below depicts two
such formats.
In Figure below, binary symbol 1 is represented by a pulse of constant amplitude for the
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duration of one bit, and symbol 0 is represented by switching off the pulse for the same
duration. This format is called nonreturn-to-zero unipolar signal, or on-off signal.
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In Figure below, symbols 1 and 0 are represented by pulses of positive and negative
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amplitude, respectively with each pulse occupying one complete bit duration. This
second format is called a nonreturn-to-zero polar signal.
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4. Regeneration:
The most important feature of PCM systems lies in the ability to control the effects of
distortion and noise produced by transmitting a PCM wave through a channel.
Shape of the pulse is affected by two mechanism:
1. Unwanted electrical noise or other disturbances.
2. Nonideal Transfer function of the transmission medium.
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This capability is accomplished by reconstructing the PCM wave by means of a chain of
regenerative repeaters located at sufficiently close spacing along the transmission route.
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Figure below show the block diagram of regenerative repeater.
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introduced into the regenerated pulse position, thereby causing distortion.
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5. Decoding:
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The first operation in the receiver is to regenerate (i.e., reshape and clean up) received
pulses.
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These clean pulses are then regrouped into code-words decoded (i.e., mapped back) into a
quantized PAM signal.
The decoding process involves generating a pulse the amplitude of which is the linear
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sum of the pulses in the code-word, with each pulse weighted by its place-value (2°, 21 ,
22, 23, . . .) in the code.
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6. Reconstruction:
Assuming that the transmission path is error-free, the recovered signal includes no noise
with the exception of the initial distortion introduced by the quantization process.
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Figure above illustrates the concept of time-division multiplexing for three bit PCM
signals.
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At the receiver, decommutator is required to sort out various signals.
If the decommutator in the receiver not synchronized to that in the transmitter, then the
messages can be interchanged. This could be most annoying in a telephone conversation;
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if we were talking to one person and the response was from another person.
In data communication system, loss of such synchronization could prove fatal. It can be
seen from Figure that there are 3 X 3 = 9 information bits in each frame and a 10th bit is
added for frame synchronization.
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The framing signal is a fixed pattern of l’s and 0’s in every 10th position and such an
alternating pattern of 1’s and 0’s will he rarely formed in any other positions for two or
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Thus, the width of pulses becomes too short. The process of generating and transmitting
very narrow pulses are very difficult.
Hence, in practice, it is necessary to restrict the number of independent message sources
that can be included within frame.
3. CD laser disks
4. Voice mail
5. Digital Video etc.
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3. Efficient SNR and bandwidth trade-off
4. Uniform format
5. Secure
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Advantages of PCM (Detailed):
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1. Low Noise Susceptibility: The PCM signal is a digital waveform. Digital waveforms
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are less susceptible to interference and noise than analog signals. This is because a digital
waveform does not have to reproduce the exact data being transmitted. A transmitted
pulse that is close enough to the expected value of a binary one can be reliably
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reproduced into a binary one. This low noise susceptibility allows PCM signals to
transmit farther than analog signals without signal degradation, information loss,
and distortion.
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2. Repeatability: A PCM signal can be received by a repeater device that decodes the data
and retransmits it. This allows PCM signals to be sent very long distances without data
corruption. Repeaters must be placed close enough to the signal source so that extreme
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noise does not corrupt the signal. Noise does not accumulate even after many passes
through multiple repeaters. This is because the signal is completely regenerated by each
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technology. The audio portion of a DVD movie is encoded using PCM with a sampling
rate as high as 192 thousand samples per second. This PCM stream can be piped directly
to an amplifier using a digital audio cable, where it is then decoded into an audible signal.
4. Encoded Signal: A PCM signal can be modulated in such a way that only a specific
decoder can make sense of the underlying data. This is useful when the data being sent
requires a level of security. The transmitter and receiver each have circuitry that is
analogous to a dictionary. This circuit maps the binary pulse-codes to their definitions.
When a pulse-code is received, the receiver looks up the meaning in the dictionary.
Anyone who intercepted the PCM signal would be left with meaningless binary data.
Disadvantages of PCM:
1. Complex Circuitry
2. Requires large bandwidth
3. Synchronization is required between transmitter and receiver
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Quantization:
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Definition1: The process of transforming sampled amplitude values of a message signal
into a discrete amplitude value is referred to as Quantization.
Definition2: In digital signal processing, quantization is the process of approximating a
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continuous range of values (or a very large set of possible discrete values) by a relatively-
small set of discrete symbols or integer values.
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The quantization Process has a two-fold effect:
1. The peak-to-peak range of the input sample values is subdivided into a finite set
of decision levels or decision thresholds that are aligned with the risers of the
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staircase, and
2. The output is assigned a discrete value selected from a finite set of representation
levels that are aligned with the treads of the staircase.
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and quantized with 16 bits (2 bytes) which can be one of 65,536 (i.e. 216) possible values
per sample.
A quantizer is memory less in that the quantizer output is determined only by the value of
a corresponding input sample, independently of earlier analog samples applied to the
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input.
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Quantization Process can be Classified on the basis of step size as:
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1. Uniform quantization: Here step size remains constant through the input range.
2. Non-uniform quantization: Here step size varies according to the input signal
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Mid-Tread type Quantizer:
It has odd number of Quantization levels.
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In the stair case like graph, the origin lies the middle of the tread portion
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In figure let us assume that the input to the quantizer x (nTs) varies from - 4Δ to + 4Δ.
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Thus, the peak to peak value of x (nT s) will be between - 4Δ to + 4Δ. Here ‘Δ’ is the step
size.
Thus, input x (nTs) can take any value between - 4Δ to + 4Δ.
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Now, the fixed digital levels are available at . These levels are
available at quantizer because of its characteristics.
Hence, according to figure above
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Thus, it may he observed from figure (b) that maximum quantization error would be ± .
Thus quantization error may be expressed as
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-------- (1)
Scenario:
Let , the quantizer will assign any one of the nearest binary levels i.e.,
either or .
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If , then
If , then .
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Thus, from above situation we see that maximum quantization error will be .
Hence Maximum quantization error will be,
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-------- (2)
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Quantization Noise/Error in PCM:
Here we derive an expression for quantization noise (i.e., error) in a PCM system for
linear quantization or uniform quantization.
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Because of quantization, inherent errors introduced in the signal. This error is called
quantization error. The quantization error is given as
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------- (1)
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Assuming that the input x(nT s) to a linear or uniform quantizer has continuous amplitude
in the range -xmax to +xmax.
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From figure above, it may be observed that the total excursion of input x(nT s) is mapped
into 'q' levels on vertical axis.
This means that when input is 4Δ, output is 7Δ/2 and when input is - 4 Δ, output is -7Δ/2.
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Thus, +xmax represents 7Δ/2 and - xmax represents -7Δ/2.
Therefore, the total amplitude range is given by,
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----- (2)
Now, if this total amplitude range is divided into 'q' levels of quantizer, then the step size
' Δ ' will be,
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------- (3)
Again, now if signal x(t) is normalized to minimum and maximum values equal to 1, then
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Now, if step size 'Δ' is considered as sufficiently small, then it may be assumed that the
quantization error will be an uniformly distributed random variable. We know that the
maximum quantization error is given as,
------- (5)
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Figure (a) above shows a uniformly distributed random variable 'X' over an interval (a,
b).
PDF of uniformly distributed random variable 'X' is given as
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-------- (6)
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Thus, with the help of above equation, the probability density function (PDF) for
quantization error ` ' may be defined as
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------- (7)
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Also, from figure (b), it may be observed that quantization error `ε' has zero average
value. In other words, the mean 'mε ' of the quantization error is zero.
Now, the signal to quantization noise ratio of the quantizer is defined as,
-------- (8)
If type of signal at input i.e., x(t) is known, then it is possible to calculate signal power.
The noise power is expressed as,
------- (9)
Here,
= the mean square value of noise voltage.
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As, here noise is defined by random varible 'ε', and PDF fε(ε) therefore, its mean square
value is given as,
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--------- (10)
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The mean square value of a random variable 'ε' is expressed as,
ES --------- (11)
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Using equation (7), above equation may be written as,
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Thus,
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------- (12)
----------- (13)
Hence, above is the value for Normalized noise power or Quantization noise power Or
Quantization error (in terms of power) for linear quantization.
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But, In PCM the normalized noise power is Δ2/12.
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Therefore,
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--------- (1)
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Assuming that input x(nTs) to a linear quantizer has continuous amplitude in the range -
xmax to +xmax. Therefore, total amplitude range is given as
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-------- (2)
------- (3)
Relation between the number of bits v and quantization levels is given as,
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-------- (4)
Now, substituting the value of q from equation (2) in equation (4), we get
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--------- (5)
---------- (6)
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This is the required relation for signal to quantization noise ratio for linear quantization in
a PCM system.
This expression shows that signal to noise power ratio of quantizer increases
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exponentially with increasing bits per sample.
Now, if we assume that inpur x(t) is normalized, i.e., xmax = 1. Also, if the destination
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signal power 'P' is normalized, i.e., P ≤ 1. Then, signal to quantization noise ratio will be,
-------- (7)
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As, xmax = 1 and P≤1, the signal to noise ratio given by equation (8) is said to be
normalized.
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-------- (8)
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Thus, signal to quantization noise ratio for normalized values of power P and amplitude
of input x(t) is given as above.
Expression for Signal to Quantization Noise Ratio for PCM that Employs
Linear Quantization assuming the PCM system has a Sinusoidal Signal input:
Assuming the Sinusoidal Signal input is a voltage signal having a peak voltage of A m.
Power of the sinusoidal voltage signal is given as,
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Here, , thus ,
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---------- (1)
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Consider a case where R =1 and power P is normalized, then normalized power is given
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as,
------ (2)
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Signal to quantization noise ratio is given as,
------ (3)
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--------- (3)
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Let Quantizer input x represents the sampled value of random variable X with zero mean
and variance .
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The Quantizer is assumed to be uniform, symmetric and mid tread type.
Let xmax denotes absolute value of the overload level of the Quantizer and Δ represent
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Step size.
Then number of Quantization level L is given by
ES ---------- (1)
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For a binary code with a code-word of n bits we can have upto 2n representation level. As
number of representation level for midtread quantizer is odd, thus L is given as below,
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-------- (2)
-------- (3)
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The ratio is called the loading factor. To avoid significant overload distortion,
the amplitude of the Quantizer input x extend from to , which corresponds to
loading factor of 4. Thus with we can write equation (3) as
------- (4)
------ (5)
------ (6)
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Hence expressing SNR in db
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------ (7)
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This formula states that each bit in codeword of a PCM system contributes 6db to the
signal to noise ratio.
ES
The equation (7) gives a good description of the noise performance of a PCM system
provided that the following conditions are satisfied.
1. The Quantization error is uniformly distributed.
TH
2. The system operates with an average signal power above the error threshold so that
the effect of channel noise is made negligible and performance is there by limited
essentially by Quantization noise alone.
DA
3. The Quantization is fine enough (say n>6) to prevent signal correlated patterns in the
Quantization error waveform.
4. The Quantizer is aligned with input for a loading factor of 4.
In a PCM system, Bandwidth B = nW or [n=B/W] substituting the value of ‘n’ we get,
U
GH
--------- (8)
RA
------ (1)
Here,
q= total number of digital levels of a q-level quantizer.
E.x., if v = 4 bits, the total number of levels will be, q = 2 4 = 16 levels.
Each sample is converted to 'v' binary bits. i.e., Number of bits per sample is v.
We know that, Number of samples per second are represented as fs. Therefore, Number
of bits per second are expressed as
P
----- (2)
----- (3)
G
Here, the number of bits per second is known as signaling rate of PCM and is denoted
H
by `r' and is given by,
ES
------ (4)
Here, fs ≥ 2 fm.
TH
Also, bandwidth needed for PCM transmission is half of the signaling rate therefore,
Transmission Bandwidth in PCM,
DA
------ (5)
U
-------- (6)
As fs ≥ 2 fm, Thus,
RA
------ (7)
Robust Quantization:
P
As the quantization error is given as,
G
---- (1)
Since, step size 'Δ' is expressed as,
H
ES
Say let x(t) is normalized, its maximum value i.e, xmax= 1. Therefore, step size 'Δ' is,
TH
---- (2)
DA
E.x.: Let consider a PCM system in which we take v = 4 bits. Then number of levels q
will be,
U
GH
Thus, we see that the quantization error is 1/16th part of the full voltage range.
Say if we assume that full range voltage is 16 volts. Then maximum quantization error
will be 1 volt.
But, for the low signal amplitudes like 2 volts, 3 volts etc., the maximum quantization
error of 1 volt which is quite high i.e., about 30 to 50%. This means that for signal
amplitudes which are close to 15 volts, 16 volts etc., the maximum quantization error
(which is same throughout the range) of 1 volt can be considered to be small.
In fact, this problem arises because of uniform quantization. Therefore non-uniform
P
quantization should be used in such cases.
G
As speech and music signals are characterized by large crest factor. This means that for
such signals the ratio of peak to rms value is quite high. Which is given as below,
H
----- (1)
ES
TH
The signal to noise ratio is given by,
------ (2)
DA
----- (3)
RA
------ (4)
----- (5)
P
When we normalize the signal x(t), then xmax = 1 Substituting this value of xmax in
G
equation (5), we get
H
----- (6)
ES
For a large crest factor of voice (i.e., speech) and music signals, P should be very very
less than one in above equation. i.e.,P << 1 for large crest factor
Therefore, actual signal to noise ratio would be significantly less than the value which is
TH
given by equation (3) since in this equation P = 1. Again, consider equation (2).
DA
---- (7)
U
This equation illustrates that the signal to noise ratio for large crest factor signal (P << 1)
GH
would be very very less than that of the calculated theoretical value. The theoretical value
is obtained for normalized power (P = 1) by equation (3).
Nonuniform Quantization:
RA
Definition: If the quantizer characteristics is nonlinear and the step size is not constant
instead if it is variable, dependent on the amplitude of input signal then the quantization
is known as nonuniform quantization.
In non-uniform quantization, the step size is reduced with the reduction in signal level.
For weak signals (P < < 1), the step size is small, therefore the quantization noise
reduces, to improve the signal to quantization noise ratio for weak signals. The step size
is thus varied according to the signal level to keep the signal to noise ratio adequately
high. This is nonuniform quantization.
The non-uniform quantization is practically achieved through a process called
companding.
P
It is required to be implemented to improve the signal to quantization noise ratio of weak
signals.
G
The quantization noise is given by
H
ES
From above equation we see that in the uniform quantization, once the step size is fixed,
the quantization noise power remains constant. However, the signal power is not
constant. It is proportional to the square of signal amplitude.
TH
Hence signal power will be small for weak signals, but quantization noise power is
constant. Therefore, the signal to quantization noise for the weak signals is very poor.
This will affect the quality of signal.
DA
The remedy is to use companding. Companding is a term derived from two words i.e.,
compression and expansion as under:
U
GH
P
Advantages of Non – Uniform Quantization :
1. Higher average signal to quantization noise power ratio than the uniform quantizer when
G
the signal pdf is non uniform which is the case in many practical situation.
2. RMS value of the quantizer noise power of a non – uniform quantizer is substantially
proportional to the sampled value and hence the effect of the quantizer noise is reduced.
H
Types of compressor characteristics:
ES
There are two type of compressor characteristics based on logarithmic compression laws
they are:
TH
1. µ - law Companding and
2. A – law Companding.
DA
μ-law:
In this companding, the compressor characteristics is defined by equation below,
U
---- (1)
GH
The μ-law is used for PCM telephone systems in the USA, Canada and Japan.
A practical value for μ is 255.
A-law:
In A-law companding the compressor characteristics is defined by equation below,
P
---- (2)
G
The normalized form of A-law compressor characteristics is shown in the figure below,
H
ES
TH
U DA
Formulas
2) Number of bits
P
3) Sampling rate
G
Where ‘W’ is the highest frequency of message signal
H
5) Transmission bandwidth
ES
TH
DA
6) Bit duration
U
GH
7) Bit rate
8) Sampling frequency
RA
9) Message bandwidth
Note:
Where
P
11)
G
H
12) Quantisation noise power or quantisation error or normalised noise power
Note: 1) ∂ or
ES
TH
2)
U DA
16)
P
and
G
18) when compression parameter ‘μ’ is given by
H
ES
19) Maximum quantisation error for an uniform quantiser is given by
TH
DA
22)
Problems:
1. The output signal to noise ratio of a 10 bit PCM was found to be 30 dB. The desired SNR
is 42 dB. It was decided to increase the SNR to the desired value increasing the number of
quantization levels. Find the fractional increase in transmission bandwidth required for
this increase in SNR.
Solution: (i) To obtain number of bits for 42 dB .
Signal to noise ratio of PCM is given as,
P
(S/N) = (4.8 + 6v) dB
G
Above equation shows that signal to noise ratio increases by 6 dB with ev bit. It is given that
H
for 10 bits
ES
The desired signal to noise ratio is 42 dB. Hence rise in ratio is 42-30 = 12 dB N
TH
We know that ratio increases by 6 dB for 1 bit. Hence 2 bits are required t ) increase signal
to noise ratio by 12 dB. Hence,
DA
= v
(10 bits) = × =5
RA
(12 bits) = = × =6
2. A telephone signal with cut-off frequency of 4 kHz is digitized into 8 bit PCM, sampled
at Nyquist rate. Calculate baseband transmission bandwidth and quantization S/ N ratio.
= vW = 4k x 8 = 32 kHz
telephone signal is non sinusoidal signal. Its signal to quantization noise ratio is given in
P
equation as,
G
4.8 + 6v
= 4.8 + 6x 8 = 52.8 dB.
H
3. A telephone signal bandlimited to 4 kHz is to be transmitted by PCM.The signal to
ES
quantization noise is to be atleast 40 dB. Find the number of levels into which signal has to
be encoded. Also find the bandwidth of transmission.
Solution:
TH
Given = 40 dB
W = 4 kHz
n number of levels (q)
DA
40 = 4.8 + 6 v
v = 5.866 or v = 6 bits.
GH
4. Show that for μ = A, the μ -law and theA-law have the same companding gain.
Solution:
The μ-law is
where u and v are the input and output of the compander respectively.
Companding gain,
P
For large values of μ,we can write
G
H
hence, ≫
the A –law is
ES
TH
U DA
5. A Television signal having a bandwidth of 4.2 MHz is transmitted using binary PCM
system. Given that the number of quantization levels is 512. Determine:
(i) Code word length
RA
(1) We know that the number of bits and quantization levels are related in binary PCM as
under:
P
G
Simplifying, we get, v = 9 bits
H
Hence, the code word length is 9 bits.
ES
(ii) We know that the transmission channel bandwidth is given as,
Thus,
GH
or
Substituting this value of in equation (i) for signaling rate, we get
RA
But
Therefore,
P
G
H
6. The bandwidth of an input signal to the PCM is restricted to 4 kHz. The input signal
ES
varies in amplitude from - 3.8 V to + 3.8 V and has the average power of 30 mW. The
required signal to noise ratio is given as 20 dB. The PCM modulator produces binary
output. Assuming uniform quantization,
TH
i. Find the number of bits required per sample.
ii. Outputs of 30 such PCM coders are time multiplexed. What would be the minimum
required transmission bandwidth for this multiplexed signal?
DA
Solution:
The given value of signal to noise ratio is 20 dB.
This means that.
U
GH
Hence,
(i) We know that the signal to quantization noise ratio is given as,
RA
Solving, we get
P
G
Since there are 30 PCM coders which are time multiplexed, the transmission bandwidth must
H
be,
ES
We also know that the signaling rate is two times the transmission bandwidth, i.e.
TH
Signaling rate, r = 840 x 2 bits/sec = 1680 bits/sec. Ans.
Thus,
= ± 0.1% = ± 0.001
We know that the maximum quantization error for an uniform quantizer is expressed as,
Or
P
Here, given
G
Substituting, values of A and xmax in equation (i), we get
H
ES
Hence, the number of levels are 10,000.
TH
i. The maximum frequency in the signal is given as 100 Hz, i.e.,
By sampling theorem minimum sampling frequency should be,
DA
ii. We know that minimum 10,000 levels should be used to quantize the signal. If binary
PCM is used, then number of bits for each samples may be calculated as under, i.e.
U
GH
Here
RA
Thus, 10,000 =
.
iv) The transmission bandwidth for PCM is expressed as, m
P
8. Twenty four voice signals are sampled uniformly and then have to be time division
G
multiplexed. The highest frequency component for each voice signal is equal to 3.4 kHz.
Now
(i) If the signals are pulse amplitude modulated using Nyquist rate sampling, what
H
would be the minimum channel bandwidth required.
(ii) If the signals are pulse code modulated with an 8 bit encoder, what would be the
ES
sampling rate? The bit rate of system is given as 1.5 x bits/sec.
Solution:
(i) As a matter of fact, if N channels are time division multiplexed, then minimum transmission
TH
bandwidth is expressed as,
DA
r = 1.5 x bits/sec
Since there are 24 channels, the bit rate of an individual channel is,
RA
Further, since each sample is encoded using 8 bits, the samples per second will be,
Note that the samples per seconds is nothing but sampling frequency
Thus, we have ,
P
9. A PCM system uses a uniform quantizer followed by a 7-bit binary encoder. The bit rate
of the system is equal to 50 x bits/sec.
G
(i) What is the maximum message signal bandwidth for which the system operates
satisfactorily?
(ii) Calculate the output signal to quantization noise ratio when a full load sinusoidal
H
modulating wave of frequency 1 MHz is applied to the input.
ES
Solution:
(i) Let us assume that the message bandwidth be Hz. Therefore sampling frequency should
be,
TH
The number of bits given as v = 7 bits
We know that the signaling rate is given as,
DA
Or
U
50 x
(ii) The modulating wave is sinusoidal. For such signal, the signal to quantization noise ratio is
expressed as,
P
(ii) What is the minimum sampling rate and what is the resulting bit transmission
rate?
G
Solution:
(i) Since the number of quantization levels given here are M = 16,
q = M = 16
H
We know that the bits and levels in binary PCM are related as,
q=
ES
Here, v = number of bits in a codeword
Thus, 16 =
Or v = 4bits
TH
(ii)again since
By sampling theorem, we know that
U DA
11. A signal having bandwidth equal to 3.5 kHz is sampled, quantized and coded by a PCM
system. The coded signal is then transmitted over a transmission channel of supporting a
transmission rate of 50 k bits/sec. Determine the maximum signal to noise ratio that can be
obtained by this system. The input signal has peak to peak value of 4 volts and rms value of
0.2 V.
Solution:
i.e.,
P
We know that the signaling rate is given by
G
H
Substituting values of r = 50 x bits/sec and 7x Hz in above equation, we get
ES
7x
Simplifying, we get
TH
8 bits
DA
The rms value of the signal is 0.2 V. Therefore the normalized signal power will be,
Normalized signal power
U
i.e
GH
12. A signal x(t) is uniformly distributed in the range ± • Evaluate maximum signal to
noise ratio for this signal.
Solution:
Given that the signal is uniformly distributed in the range ± xmax, therefore we can write its PDF
(using the Standard Uniform Distribution) as under:,
P
G
The mean square value of random variable X is given as ,
H
ES
TH
U DA
GH
P
G
We know that the relation between step size, maximum amplitude of signal and number of levels
is given as
H
Step size
ES
TH
Therefore, normalized signal power,
DA
or
P
13. Given an audio signal consisting of the sinusoidal term given as
(i) Determine the signal to quantization noise ratio when this is quantized using 10
G
bit PCM.
(ii) How many bits of quantization are needed to achieve a signal to quantization
H
noise ratio of atleast 40 dB?
Solution:
ES
Here,
TH
This is sinusoidal signal applied to the quantizer.
(i) Let us assume that peak value of cosine wave defined by x(t) covers the complete range of
quantizer.
i.e., covers complete range
DA
It is expressed as
U
V=10v
Thus,
RA
ii) For sinusoidal signal, again, let us use the same relation
i.e ,
To get signal to noise ratio of at least 40 dB we can write above equation as,
Hence, at least 7 bits are required to get signal to noise ratio of 40 dB. Ans.
14. A 7 bit PCM system employing uniform quantization has an overall signaling rate of 56
P
k bits per second. Calculate the signal to quantization noise that would result when its
input is a sine wave with peak amplitude equal to 5 Volt. Find the dynamic range for the
G
sine wave inputs in order that the signal to quantization noise ratio may be less than 30
dBs. What is the theoretical maximum frequency that this system can handle?
H
Solution:
The number of bits in the PCM system are
ES
v = 7 bits
Assume that 5 V peak to peak voltage utilizes complete range of quantizer. Then, we can find the
signal to quantization noise ratio as,
TH
DA
56 x =7•
Simplifying, we get Sampling frequency, fs = 8 x 103 Hz
GH
15. Given a sine wave of frequency fu, and amplitude Am applied to a delta modulator having step size 4.
Show that the slope overload distortion will occur if
P
It may be noted that the slope of x(t) will be maximum when derivative of x(t) with respect to `t' will be
maximum. The maximum slope of delta modulator may be given as,
G
H
ES
We know that, slope overload distortion will take place if slope of sine wave is greater than slope of delta
modulator i.e.,
TH
U DA
GH
RA
Hence proved.
15. Determine the output signal to noise ratio of a linear delta modulation system for a 2
kHz sinusoidal input signal sampled at 64 kHz. Slope overload distortion is not present and
the post reconstruction filter has a bandwidth of 4 kHz.
Solution :
We know that
Here,
= 64 kHz, = 2 kHz and = 4kHz
Therefore,
P
G
16. For the same sinusoidal input of example 4.15, calculate the signal to quantization noise
ratio of a PCM system which has the same data rate of 64 kbits/s. The sampling frequency
H
is 8 kHz and the number of bits per sample is N = 8. Comment on the result.
Solution :
ES
The signal to noise ratio of a PCM system is given by,
PCM.
17. A binary channel with bit rate r= 36000 bits per second (b/s) is available for PCM voice
transmission .Evaluate the appropriate values of the sampling rate , the quantizing level
U
Therefore, we have, v 5,
and also,
and
18. An analog signal is sampled at the Nyquist rate and quantized into q levels. Find the
time duration τ of 1 bit of the binary-encoded signal.
Solution:
where indicates the next higher integer to be taken if is not integer value i.e,
binary pulses must be transmitted per second.
P
G
where is the Nyquist interval.
H
19. The output signal-to-quantizing-noise ratio (SNR)0 in a PCM system is defined as the
ratio of average signal power to average quantizing noise power. For a full-scale sinusoidal
ES
modulating signal with amplitude A, prove that
TH
Where q is the number of quantizing levels.
DA
Solution:
Since, here peak-to-peak excursion of the quatizer input is 2A. Therefore, the quantizer step size
will be
U
GH
The output signal-to-quantizing-noise ratio of a PCM system for a full scale test tone is,
therefore,
P
In a binary PCM system, q = , where v is the number of binary digits. Then, we have
G
H
Now,since
ES
TH
Therefore ,
DA
Thus we have ,
U
GH
Then, the number of levels required is q = = 128, and corresponding output signal-to-
quantizing noise ratio will be
NOTE: Equation W indicates that each bit in the code word of a binary PCM system contributes
6 dB to the output signal.to-quantizing noise ratio. In fact, this is called the 6 dB rule.
21. Consider an audio signal with spectral components limited to the frequency band of 300
to 3300 Hz. A PCM signal is generated with a sampling rate of 8000 samples/s. The
required output signal-to-quantizing-noise ratio is 30 dB.
(i) What is the minimum number of uniform quantizing levels needed, and what is the
minimum number of bits per sample needed?
P
(ii) Calculate the minimum system bandwidth required.
(iii) Repeat parts (i) and (ii) when a g-law compander is used with g = 255.
G
Solution:
(i) Here, we have
H
ES
TH
DA
We have
P
G
The minimum number of bits per sample is 7.
The minimum bandwidth required for this case will be
H
ES
22. Consider a sinusoidal signal m(t) = A cos wmt applied to a delta modulator step size A.
Show that the slope overload distortion will occur if
TH
DA
We have so that
GH
RA
23. A DM system is designed to operate at 3 times the Nyquist rate for a signal with a 3 kHz
bandwidth. The quantizing step size is 250 mV.
(i) Determine the maximum amplitude of a 1-kHz input siinusoid for which the delta
modulator does not show slope overload.
(ii) Determine the posfiltered output signal-to-quantizing-noise ratio for the signal of
part (i)
Solution:
We have
P
G
H
The maximum allowable amplitude of the input sinusoid is
ES
TH
DA
ii) Assuming that the cutoff frequency of the low-pass filter is I'm, we have
U
GH
24. The pulse rate in a DM system is 56,000 per sec. The input signal is
RA
Find the minimum value of step size which will avoid slope overload distortion. What
would be the disadvantages of choosing a value of larger than the minimum?
Solution:
Input signal,
P
G
H
ES
TH
Hence, larger step size out of two will be the required step size. i.e., = 0.56 V. If a value larger
than the minimum will be choosen, then granular noise will occur.
DA
25. Bandwidth of the input to pulse code modulator is restricted to 4 kHz. The input varies
from - 3.8 V to 3.8 V and has the average power of 30 mW, the required signal to
U
quantization noise power ratio is 20 dB. The modulator produces binary output. Assume
uniform quantization Calculate the number of bits required per sample.
GH
Solution:
Given that
RA
P
G
H
ES
Hence, n = 7 = number of bits required per sample.
TH
26. A low pass signal of 3 kHz bandwidth and amplitude over - 5 Volts to + 5 Volts range is
sampled at Nyquist rate and converted to 8-bit PCM using uniform quantization. The
DA
and (iii).
Solution :
GH
Given that
RA
Therefore,
P
Substituting above value of A in equation (i), we obtain
G
H
Now, let us calculate the bit transmission rate (r).
The bit transmission rate or signaling rate is the number of bits transmitted by the PCM system
ES
per second.
Therefore,
TH
As the signal is sampled at Nyquist rate,
we have
DA
Therefore,
27. For a full scale sinusoidal modulating signal with peak value A, show that, output signal
to quantization noise ratio in binary PCM system is given by,
P
where M= Number of quantization levels.
A compact disc recording system samples each of the two-stereo signals with a 16 bit A/D
G
converter at 44.1 Kb/sec.
(i) Determine output S/N ratio for a full scale sinusoid.
(ii) The bit stream of digitized data is augmented by addition of error correcting bits,
H
clock extraction bits etc. and these additional bits represent 100% overhead.
Determine output bit rate of CD system.
ES
(iii) The CD can record an hour's worth of music. Determine number of bits recorded
on CD.
Solution :
TH
There are two stereo channels.
DA
(i) Output signal to noise ratio for full scale sinusoid is given by
U
GH
ii) Now, let us evaluate the output bit rate of the CD system.
The bit rate for each of two stereo channels =
RA
Including the additional 100% overhead, the output bit rate will be
28. Determine the output SNR in a DM system for 1 kHz sinusoid, sampled at 32 kHz
without slope overload and followed by a 4 kHz post construction filter. Derive the formula
used.
P
Solution :
Given that,
G
H
It is given that there is no slope overload.
The output signal to noise ratio in a DM system is expressed as
ES
TH
Therefore,
DA
29. The bandwidth of TV video plus audio signal is 4.5 MHz. If this signal is converted into
U
PCM bit stream with 1024 quantization levels, determine number of bits/sec of the
GH
resulting signal. Assume that the signal is sampled at the rate 20% above Nyquist rate.
Solution :
Given that
RA
Therefore, bits/sec .
30. If a voice frequency signal is sampled at the rate of 32,000 samples/sec and
P
characterized by peak value of 2 Volts, determine the value of step size to avoid slope
overload. What is quantization noise power Ng and corresponding SNR ? Assume
bandwidth of signal as 4 kHz.
G
Solution :
Given that
H
Peak value of the signal . Bandwidth .
i) Step size A to avoid slope overload can be calculated as under :
ES
To avoid slope overload the following condition must be satisfied :
TH
Substituting the values, we obtain
U DA
GH
31. A compact disc (CD) records audio signals digitally by PCM. Assume audio signal's
bandwidth to be 15 kHz. If signals are sampled at a rate 20% above Nyquist rate for
practical reasons and the samples are quantized into 65,536 levels. Determine bits/sec
required to encode the signal and minimum bandwidth required to transmit encoded
P
signal.
Solution :
Given that
G
H
Signaling rate (r) can be calculated as under :
We know that
ES
TH
DA
32. In a single integration DM scheme, the voice signal is sampled at a rate of 64 kHz. The
maximum signal amplitude is 1 Volt.
(i) Determine the minimum value of step size to avoid slope overload.
(ii) Determine granular noise power No, if the voice signal bandwidth is 3.5 kHz.
(iii) Assuming signal to be sinusoidal, calculate signal power So and signal to noise
ratio (SNR).
(iv) Assuming that the voice signal amplitude is uniformly distributed in the range (- 1,
1), determine So and SNR.
Solution :
Given that
P
G
H
ii)
ES
Granular noise power is expressed as
TH
Solving, we get
DA
Question Bank:
U
1. What are the advantages of digital representation of analog signals? December 2011 (04
M)
GH
2. What is the need for non-uniform quantization? Explain the µ-law companding.
December 2011 (09 M), December 2013 (08 M)
3. Derive an expression for the SNR of a PCM system. December 2011 (07 M)
RA
4. Derive an expression for output SNR of the quantizer and show that
in decibles if a sinusoidal signal is quantized. December 2013 (08 M)
5. For a binary PCM signal, determine ‘L’ if the compression parameter and the
minimum . Determine the with this value of L.
December 2013 (04 M), June 2012 (06 M)
6. With a neat block diagram and waveform, explain TDM. June 2012 (06 M)
7. Show that the signal to quantization noise power ratio of a uniform quantizer is
P
uniform quantization, each bit in the codeword of a PCM contributes 6 dB to SNR. June
2013 (08 M)
G
11. Six independent message sources of bandwidths w, w, 2w, 2w, 3w and 3w hertz are to be
transmitted on TDM. Set up a scheme to accomplish this requirement, with each message
signal sampled at its Nyquist rate. June 2013 (05 M)
H
12. The signal m(t) = 6 sin (2πt) Volts, is transmitted using 4-bit binary PCM system. The
quantizer is of midriser type with a step size of 1 Volt. The sampling frequency is 4 Hz
ES
with samples taken at t = ±1/8, ±3/8, ±5/8, … sec. Sketch the PCM wave for one
complete cycle of the input. June 2013 (07 M)
13. Explain regenerative repeater in a PCM system with a block diagram. June 2014 (05 M)
TH
14. The bandwidth of a signal is 3.4 kHz. If the signal is converted to PCM bit stream with
1024 levels, determine the number of bits per second generated by the PCM system.
Assume that the signal is sampled at the rate of 20% above the Nyquist rete. June 2014
DA
(06 M)
15. Derive an expression for the output SNR of s uniform quantizer in terms of step size of
the quantizer. Hence show that for mid-tread type uniform quantizer the SNR is
U
(SNR)output = 6n – 7.2 dB, whrer ‘n’ is the number of bits per sample. Assume a loading
factor of 4. June 2014 (12 M)
GH
RA