SIGNALS SPECTRA REVIEWER

Signal is a single-valued function of time that conveys information.

A signal is a description of how one parameter varies with another parameter. For instance, voltage
changing over time in an electronic circuit, or brightness varying with distance in an image.

Continuous signals are usually represented with parentheses, while discrete signals use brackets.

the input signal is called: x(t) or x[n], while the output is called: y(t) or y[n]

A system is any process that produces an output signal in response to an input signal.

It may:

• involve signal processing, • it can be linear/non-linear and

• also include software.
Algorithm is the method or set of rules for implementing the system by a program that performs the
corresponding mathematical operations.

TYPES OF DOMAIN

Time Domain simply means that the independent variable is measured in time.

The frequency domain is exactly the same as the time independent variable except that the frequency
becomes the independent variable.

Spectra describes the frequency content of the signal.

Spatial domain - is exactly the same as the time domain description, except that the spatial becomes the
independent variable. Spatial typically describes a distance.

Signal processing - is extracting information from a signal, conditioning signal for subsequent use, signal
transformation, or altering a signal structure.

ELEMENTS OF SIGNAL PROCESSING

Analog Signal Processing - the signal is processed directly in its analog form.

Digital Signal Processing - the signal is converted to digital before processing and converted back to
analog.

ADVANTAGE OF DSP OVER ANALOG:

• Flexibility
• Accuracy
 • Cheaper
• Easy storage LIMITATIONS:

1. speed of operation

2. bandwidth considerations

DIGITAL SIGNAL PROCESSING (DSP ) is the study of signals in a digital representation and the
processing methods of these signals. The algorithms required for DSP are sometimes performed
using specialized computers, which make use of specialized microprocessors called digital signal
processor.

DSP is the mathematics, the algorithms, and the techniques used to manipulate these signals after they
have been converted into a digital form.

APPLICATIONS OF DSP:

• audio signal processing

• audio and video compression
• digital image processing
• speech processing and speech recognition
• digital communication
Most DSP techniques are based on a divide-and conquer strategy called superposition.

Classification of Signals

ACCORDING TO CHANNEL:

Single Channel - generated by single source.

Multiple Channel - generated by multiple sources or multiple sensors

ACCORDING TO DIMENSION:

ONE DIMENSIONAL - a function of single or independent variable

MULTI-DIMENSIONAL (M DIMENSIONAL) - has M-independent variables

EXAMPLES

1. S(t) = A sin 3 t
2. S(t) = t² + 4t – 8

3. S(x, y) = 4x² + 9xy – 13y³

4. S(t) = S1(t) S2(t) S3(t)

ACCORDING TO THE SIGNAL VALUE:

(a) real

(b) complex

CLASSIFICATION ACCORDING TO CHARACTERISTICS OF TIME VARIABLE AND THE VALUE THEY TAKE:

A. CONTINUOUS–TIME SIGNALS OR ANALOG SIGNALS - are defined for every value of time they
take on continuous interval (a, b), where a can be – and b can be .

B. DISCRETE–TIME SIGNALS - are defined only at certain specific values of time. These time
instants needs not to be equidistant, but they are often equally spaced intervals.

CONTINUOUS–VALUED SIGNALS - if a signal takes on all possible values on infinite or infinite

range.

DISCRETE–VALUED SIGNALS - if the signal takes on values from a finite set of possible values.

NOTE: A discrete-time and discrete-valued signal refers to digital signal.

ACCORDING TO THE SIGNAL MODEL

• DETERMINISTIC SIGNAL - a signal that can be uniquely described by an explicit mathematical

expressions, table of data or a well-defined rule. All past, present and future values of the signal
are known precisely without uncertainty.
• RANDOM SIGNAL - signals that cannot be described by explicit mathematical formulas.

Consider the example equation of harmonic oscillation formula of a signal:

Xa(t) = A cos ( t + ), – <t<

where: A → amplitude f →

frequency

→radian frequency ➔ 2 f

→phase
PROPERTIES:

(a). For every fixed value of frequency, f:

Xa(t) is periodic → Xa(t + Tp) = Xa(t)

PERIODIC SIGNALS → a signal is periodic if it completes a pattern with in a measurable time frame, and
repeats that pattern over identical subsequent periods.

APERIODIC or NON-PERIODIC SIGNALS → if the signal changes constantly without exhibiting a pattern
or cycle that repeats over time.

(b). Continuous-time sinusoidal signals with distinct frequencies are themselves distinct.

(c). Increasing the frequency, f, result in an increase in the rate of oscillation of the signal, in the sense
that more periods are included in a given time interval.

EXAMPLE:

If Xa(t) = 20 cos100 t, determine its periodic time (Tp).

Given: Xa(t) = 100 sin200 t (a)Find Tp.

(b) For Xa(t) = 100 sin400 t, Find Tp.

(c) at 50ms, how many cycles does Xa(t) of (a) and (b) will take

DISCRETE SIGNALS TIME SINUSOIDAL

Consider the example equation of a signal:

X[n] = A cos ( n + ) where:

n → nth time (any integer) PROPERTIES :

(a). A discrete-time sinusoid is periodic if its frequency, f, is a rational number: X[n

+ N] = X[n] where:

N → the smallest value of N in the fundamental period

(b). Discrete-time sinusoids whose frequencies are separated by an integer multiple of 2 are identical.

(c). The highest rate of oscillation in a discrete-time sinusoid is attained when: =

( = - ) or f = ½ (f = -½ )
HARMONICALLY RELATED COMPLEX EXPONENTIALS → these are sets of periodic complex exponentials
with fundamental frequencies that are multiples of a single positive frequency.

CONTINUOUS-TIME EXPONENTIALS:

Sk (t)= e^jk ot = e^jk2 fot (kth harmonics) where: k

→ 0, 1, 2, 3 …..

EXAMPLE:

1. Identify the whether signal is periodic or non-periodic:

a) cos 0.3 n
b) sin 56n

2. Consider the signal Xa (t) = 10 cos 60 t. Find the 6th and 10th harmonic and the fundamental period
of each.

ANALOG DIGITAL (A/D) AND DIGITAL ANALOG (D/A) CONVERSION

SAMPLING - the conversion of a continuous–time signal into a discrete-time signal

In order to properly sample an analog signal the Nyquist-Shannon sampling theorem must be satisfied.
In short, the sampling frequency must be greater than twice the bandwidth of the signal.

QUANTIZATION - the conversion of a discrete–time continuous valued signal into a discrete–time discrete
valued signal (digital).

CODING - in communication system, it’s the altering of the characteristics of a signal to make the signal
more suitable for an intended application. In DSP, it’s the process where each discrete value Xq[n] is
represent by a b–bit binary sequence.
SAMPLING OF ANALOG SIGNALS

Where:
X [n] → discrete – time signal obtained by
“taking samples” of the analog signal Xa(t) every
T seconds
Ts → sampling period or sample interval
fs → sampling rate or sample frequency
f → digital or relative or absolute frequency
F → analog frequency

X[n] = Xa[nT] = cos (2𝝅FnT + Ө)

Consider:

f = F / fs T = 1 / fs

X[n] = cos {2𝝅FnT + Ө}

X[n] = cos (2𝝅fn + Ө) - discrete time signal

But according to discrete time signal properties:

- 1⁄2 ≤ f ≤ 1⁄2 also:

NYQUIST RATE
→ the minimum sampling rate that can be used to sample on analog with frequency F.

fs(min) = 2Fmax

ALIAS FREQUENCY
→ the frequency of a signal at which when sampled at a given rate produces the same value as the
original signal.

FOLDING FREQUENCY
→ the maximum analog frequency that can be sampled by a sampling rate of fs.

F(max) = fs/2
EXAMPLES

1. Consider the two signals:

X1(t) = cos 20𝝅t X2(t) = cos 100𝝅t

For a sampling rate of fs = 40 Hz, determine their corresponding discrete–time signal.

2. Consider the signal:

x(t) = 100 cos 8000𝝅t

Determine the minimum sampling rate.

3. What is the minimum sampling rate that can be used to sample voice signal?

Note: voice frequency is 300Hz – 3000Hz

4. Consider the analog signal:

Xa(t) = 3 cos 100𝝅t

a) Determine the minimum sampling rate to avoid aliasing.

b) Suppose that the signal is sampled at a rate of 200Hz, what is the discrete-time signal obtained
after sampling?
c) Suppose that fs = 75Hz, what is the discrete–time signal obtained after sampling?
d) What is the frequency {–(fs/2) < F < (fs/2)} of a sinusoid that yields samples identical to those
obtained in (c)?
SAMPLING THEORY

The following are considered in converting a continuous–time signal into a discrete–time signal:

1. Always check for those components (of the discrete–time signal) which are identical.

2. If the signals are identical, combine them. Select the lower frequency.

3. Also simplify the discrete–time signal.

EXAMPLES

Consider the analog signal

Xa(t) = 3cos50𝝅t + 10sin300𝝅t – cos100𝝅t

a) What is the Nyquist rate for this signal?

b) What is the discrete–time signal obtained using the
Nyquist rate?

Consider the analog signal:

Xa(t) = 3cos2000𝝅t + 5sin5000𝝅t + 10cos8000𝝅t

a) What is the Nyquist rate for this signal?

b) Assume that the sampling rate is 750Hz sample/sec, determine the discrete–time signal
obtained?
c) What is the analog signal ya(t) we can reconstruct from the samples if we use ideal interpolation?
QUANTIZATION - process of approximating a continuous range of values (or a very large set of possible
discrete values) by a relatively-small set of discrete symbols or integer values.

QUANTIZATION ERROR - the error introduced in representing the continuous–valued signal by a finite
set of discrete value levels.

Let:

X[n] = discrete–time signal (continuous–valued)

Q{X[n]} = quantizer operation

Xq[n] = quantized samples

Q{X[n]} = Xq[n]

QUANTIZATION ERROR ( eq[n])

eq[n] = Xq[n] - X[n]

EXAMPLE:

Consider the signal:

a.) Obtain discrete–time signals at n=0 to 9.
b.) Truncate the values and round them according to the
quantization values at step size of 0.1.
c.) Determine the quantization error for each value.

Note:
NOTE:

For rounding quantizer:

where:

Xmax = maximum amplitude

Xmin = minimum amplitude

L = quantization levels

SIGNAL TO QUANTIZATION NOISE RATIO (SQNR)

🡪 signal to quantization noise ratio of signal power to noise power.

SQNR = 1.76 + 6.02b

where: b = number of bits used in coding stage

CODING OF QUANTIZED SAMPLES

CODING - assigning of a unique binary number to each quantization level.

where:

b = number of bits to be used

L = number of quantization levels

SAMPLE PROBLEM:

1. Determine the number of bits used for a quantization level of 11.

2. If Xa(t) = 3cos600πt + 2cos1800πt

→ operated at 10,000 bits/sec.

→ 1024 different voltage levels.

Find:

a) Sampling frequency
b) Nyquist rate
c) Folding frequency
d) Resolution
 - A digital to analog converter (DAC) is used to convert the digital signal back to analog.

- Analog electronic filters used to comply with the sampling theorem. The electronic filter placed
before an ADC is called an antialias filter. It is used to remove frequency components above
one-half of the sampling rate that would alias during the sampling. The electronic filter placed
after a DAC is called a reconstruction filter. It also eliminates frequencies above the Nyquist
rate, and may include a correction for the zeroth-order hold.

Bessel Filter

- Also known as Thompson Filter

- Good step response
- Worst roll-off rate but can be improved by adding more poles

Butterworth Filter

- Has the flattest passband

- the Butterworth filter is optimized to provide the sharpest roll-off possible without allowing
ripple in the passband

Chebyshev Filter

- Has the sharpest roll-off rate

- Has 6% ripple in the passband
- Has ringing and possibility of overshoot