NAME: Nemo, Czriss Paulimer C.

Year/Section: 3/A
Date: October 16,2020

Problem Set No 1.
Answer the following:

1. Define digital transmission.

- It is the process of transmitting digital signals from one point to another in a communication
system.
2. List and describe the advantages and disadvantages of digital transmission.
Advantages:
a. Digital signals are less susceptible to noise compared to analog signals.
b. Their noise immunity characteristic is due to the use of signal regeneration rather than
signal amplification.
c. Digital signals are also easier to process and manipulate.
d. Digital signals are much more secure since they can be encrypted.
Disadvantages:
a. If the signal is originally an analog signal, then the digital transmission employed will
require more bandwidth.
b. Analog signals that are subject to digital transmission must first be converted to digital
signals thus, requiring more equipment.
3. Briefly describe pulse width modulation, pulse position modulation, and pulse amplitude
modulation.
- In a general context, all these types of pulse modulations produce digital pulses that
conforms to the provided analog signal. Pulse Width Modulation results to discrete pulses
that have constant amplitude and each width varying according to the analog signal’s
amplitude during that instant. Pulse Position Modulation produces a pulse that is of constant
width that is placed in different positions in accordance to the amplitude of the analog signal
in a particular sample. Pulse Amplitude Modulation creates constant-width pulses that are in
the same position. Each pulse’s amplitude corresponds to the analog signal’s amplitude
during a particular sample.
4. Define and describe pulse code modulation.
- Pulse Code Modulation is a process wherein analog signals are being digitally coded. This is
done by assigning the signal voltages in each sample of the analog signal to binary numbers.
The provided figure below is an example of an analog signal that is being subjected to pulse
code modulation.
 - As seen in the figure above, each sample of the analog signal is designated with an 8-bit
number. The presence of a pulse that is of constant amplitude and width designates a logic 1
condition while its absence corresponds to a logic 0. For instance, the first sample’s signal
voltage is equal the binary number 111000102.
5. Explain flat-top and natural sampling.
- Flat-top Sampling makes use of a sample-and-hold circuit in order to periodically sample the
analog signal and convert these samples into constant-amplitude pulses. This only traces the
analog signal with the starting points of each pulse.
- Unlike flat-top sampling, The tops of the pulses generated via Natural Sampling retains the
analog signal’s shape.
6. Describe the Nyquist sampling theorem.
- The Nyquist Sampling Theorem, also known as the Sampling theorem, states that an analog
signal that is subject to Digital Signal Processing (DSP) must have a sampling frequency that
is twice the bandwidth of a given signal. It is mathematically expressed as:
f s (min)=2 B
Where:
f s (min )=minimum sampling frequency ∨the Nyquist frequency ( Hertz)
B=bandwidth (Hertz )
7. Describe folded binary codes.
- Folded Binary Codes is a type of code wherein the codes seen in the bottom half are mirror
images of those above excluding their signed bits.
8. Define and explain dynamic range.
- The Dynamic Range of a system is simply the ratio of the largest possible magnitude of the
output signal to the least possible magnitude discernible. It is mathematically expressed as
V max
DR=20 log ⁡( )
V min
Where:
DR=dynamic range
V max =maximum voltage magnitude
V min =quantum value
- The dynamic range simply indicates the range of input levels where the receiver is still
operational. The number of bits necessary in a signal’s PCM code is dependent in the
system’s dynamic ranged. Their relationship is defined as
2n−1=DR
Where:
n=number of bits
DR=dynamic range
- In a linear PCM system, the dynamic range is
DR=1.76+ 6.02m
Where:
m=number of bits per sample
9. Explain PCM coding efficiency.
- PCM Coding Efficiency is a numerical value as to how efficient a PCM code is utilized.
Mathematically, it is the ratio of the minimum number of bits utilized to acquire a specific
dynamic range to the actual number of bits used.
minimumnumber of bits(inclusive of signed bit )
coding efficiency= ×100
actual number of bits (inclusive of signed bit )
10. Describe signal-to-quantization noise ratio.
 - The Signal-to-Quantization Noise Ratio is the ratio of the received signal’s strength to the
quantization error. The maximum quantization error is equal to half the least discernible
magnitude of the received signal. In PAM waveforms and analog signals, the signal-to-
quantization ratio is not constant since it varies in amplitude. It is described mathematically
as
v2
SQR(dB)=10 log

Where:
() R
q2
( )
12
R

SQR( dB)=Signal−¿−Quantization Noise Ratio ( dB )

v=rms signal voltage ( volts )
q=quantization interval(volts)
v2
=average signal power (watts)
R
q2
( )
12
=average quantization power (watts)
R
11. What is Line Encoding? Explain Unipolar, Polar and Bipolar Line Encoding.
- Line Encoding is a process of representing digital signals wherein the magnitude change
between any successive steps is equal. Since the resolution is independent from the
amplitude of the signals, it consumes lesser bandwidth hence, efficient with power. Unipolar
Line Encoding only makes use of a single polarity, i.e. all signals are either above or below
the axis. An example is logic circuits since they only switch between two states; logic 1 or
logic 0. It is also known as a Non Return to Zero signal because it does not revert back to 0 in
the middle of a bit.
- Polar Line Encoding encompasses multiple polarities within a given signal, i.e. uses two
levels of amplitude. Unlike the unipolar, it can either be a Non Return to Zero or a Return to
Zero signal.
- The Bipolar Line Encoding utilizes three voltage levels; positive, negative and zero. An
element is assigned to the voltage level zero while the other can switch to either positive or
negative.
12. It is necessary to transmit the human voice using a frequency range from 300 Hz to 3.5 kHz
using a digital system.
(a) What is the minimum required sampling rate, according to theory?
f s (min)=2 B=2 ( 3500 Hz−300 Hz )=6400 Hz∨6.4 kHz
(b) Why would a practical system need a higher rate than the one you calculated in part (a)?
- In practice, higher rates are used since other external conditions such as channel
complexity, time constraints, availability of the practical circuits, application, etc. are
also taken into consideration. Another reason is that the theorem used in part (a) is
under the assumption that an ideal low-pass filter is used which is not the case in
practice.
13. The human voice actually has a spectrum that extends to much higher frequencies than are
necessary for communication. Suppose a frequency of 5 kHz was present in a sampler that
sampled at 8 kHz.
(a) What would happen?
- According to the Nyquist Sampling Theorem, the minimum sampling frequency of a
signal must be at least twice its bandwidth. In the case, the signal with a frequency
of 5 kHz must be sampled at a frequency of 10 kHz. However, it is only sampled at 8
kHz thus, a distortion called aliasing will be present in the resulting signal.
(b) How can the problem described in part (a) be prevented?
- In order to prevent such complications, the signal must be sampled at higher rates.
Since this may seem costly, cascading another filter in the system can also suffice.
14. A 1-kHz sine wave with a peak value of 1 volt and no dc offset is sampled every 250
microseconds. Assume the first sample is taken as the voltage crosses zero in the upward
direction. Sketch the results over 1ms using:

(a) PAM with all pulses in the positive direction

 (b) PDM

(c) PPM

15. The compact disc system of digital audio uses two channels with TDM. Each channel is sampled
at 44.1 kHz and coded using linear PCM with sixteen bits per sample. Find:
(a) the maximum audio frequency that can be recorded (assuming ideal filters)
f s=2 f max
f s 44,100 Hz
f max = = =22,050 Hz∨22.05 kHz
2 2
(b) the maximum dynamic range in decibels
DR=1.76+ 6.02 ( 16 ) =98.08 dB
(c) the bit rate, ignoring error correction and framing bits
bit rate=f s × m=44,100 Hz ×16=705,600 bps∨705.6 kbps
(d) the number of quantizing levels
 N=2m=216=65,536
16. Determine the Nyquist sampling rate for the following maximum analog input frequencies: 2
kHz, 5 kHz, 12 kHz, and 20 kHz.
f s (2 kHz )=2 ( 2,000 Hz )=4,000 Hz∨4 kHz
f s (5 kHz )=2 ( 5,000 Hz )=10,000 Hz∨10 kHz
f s (12 kHz) =2 (12,000 Hz ) =24,000 Hz∨24 kHz
f s (20 kHz )=2 ( 20,000 Hz )=40,000 Hz∨40 kHz
17. Determine the maximum analog input frequency for the following Nyquist sample rates: 2.5
kHz, 4 kHz, 9 kHz, and 11 kHz.
2,500 Hz
f a (2.5 kHz )= =1,250 Hz∨1.25 kHz
2
4,000 Hz
f a (4 kHz)= =2,000 Hz∨2 kHz
2
9,000 Hz
f a (9 kHz )= =4,500 Hz∨4.5 kHz
2
11,000 Hz
f a (11kHz )= =5,500 Hz∨5.5 kHz
2

18. Determine the dynamic range in dB for the following n-bit linear sign-magnitude PCM codes: n
= 7, 8, 12, and 14.
DR(7) =1.76+6.02 ( 7 )=43.9 dB
DR(8) =1.76+6.02 ( 8 )=49.92 dB
DR(12)=1.76+ 6.02 ( 12 )=74 dB
DR(14) =1.76+6.02 ( 14 )=86.04 dB