Tag: hilbert transform

Phase demodulation via Hilbert transform: Hands-on

Key focus: Demodulation of phase modulated signal by extracting instantaneous phase can be done using Hilbert transform. Hands-on demo in Python & Matlab.

Phase modulated signal:

The concept of instantaneous amplitude/phase/frequency are fundamental to information communication and appears in many signal processing application. We know that a monochromatic signal of form x(t) = a cos(ω t + ɸ) cannot carry any information. To carry information, the signal need to be modulated. Different types of modulations can be performed – amplitude modulation, phase modulation / frequency modulation.

In amplitude modulation, the information is encoded as variations in the amplitude of a carrier signal. Demodulation of an amplitude modulated signal, involves extraction of envelope of the modulated signal. This was discussed and demonstrated here.

This article is part of the following books
● Digital Modulations using Matlab : Build Simulation Models from Scratch, ISBN: 978-1521493885
● Digital Modulations using Python ISBN: 978-1712321638
● Wireless communication systems in Matlab ISBN: 979-8648350779
All books available in ebook (PDF) and Paperback formats

In phase modulation, the information is encoded as variations in the phase of the carrier signal. In its generic form, a phase modulated signal is expressed as an information-bearing sinusoidal signal modulating another sinusoidal carrier signal

\[x(t) = A cos \left[ 2 \pi f_c t + \beta + \alpha sin \left( 2 \pi f_m t + \theta \right) \right]   \quad \quad \quad (1)\]

where, m(t) = α sin (2 π fm t + θ ) represents the information-bearing modulating signal, with the following parameters

α – amplitude of the modulating sinusoidal signal
fm – frequency of the modulating sinusoidal signal
θ – phase offset of the modulating sinusoidal signal

The carrier signal has the following parameters

A – amplitude of the carrier
fc – frequency of the carrier and fc>>fm
β – phase offset of the carrier

Demodulating a phase modulated signal:

The phase modulated signal shown in equation (1), can be simply expressed as

\[x(t) = A cos \left[ \phi(t) \right]    \quad\quad\quad (2) \]

Here,  ɸ(t) is the instantaneous phase  that varies according to the information signal m(t).

A phase modulated signal of form x(t) can be demodulated by forming an analytic signal by applying Hilbert transform and then extracting the instantaneous phase. This method is explained here.

We note that the instantaneous phase is ɸ(t) = 2 π fc t + β + α sin (2 π fm t + θ) is linear in time, that is proportional to 2 π fc t . This linear offset needs to be subtracted from the instantaneous phase to obtained the information bearing modulated signal. If the carrier frequency is known at the receiver, this can be done easily. If not, the carrier frequency term 2 π fc t needs to be estimated using a linear fit of the unwrapped instantaneous phase. The following Matlab and Python codes demonstrate all these methods.

Matlab code

%Demonstrate simple Phase Demodulation using Hilbert transform
clearvars; clc;
fc = 240; %carrier frequency
fm = 10; %frequency of modulating signal
alpha = 1; %amplitude of modulating signal
theta = pi/4; %phase offset of modulating signal
beta = pi/5; %constant carrier phase offset 
receiverKnowsCarrier= 'False'; %If receiver knows the carrier frequency & phase offset

fs = 8*fc; %sampling frequency
duration = 0.5; %duration of the signal
t = 0:1/fs:1-1/fs; %time base

%Phase Modulation
m_t = alpha*sin(2*pi*fm*t + theta); %modulating signal
x = cos(2*pi*fc*t + beta + m_t ); %modulated signal

figure(); subplot(2,1,1)
plot(t,m_t) %plot modulating signal
title('Modulating signal'); xlabel('t'); ylabel('m(t)')

subplot(2,1,2)
plot(t,x) %plot modulated signal
title('Modulated signal'); xlabel('t');ylabel('x(t)')

%Add AWGN noise to the transmitted signal
nMean = 0; %noise mean
nSigma = 0.1; %noise sigma
n = nMean + nSigma*randn(size(t)); %awgn noise
r = x + n;  %noisy received signal

%Demodulation of the noisy Phase Modulated signal
z= hilbert(r); %form the analytical signal from the received vector
inst_phase = unwrap(angle(z)); %instaneous phase

%If receiver don't know the carrier, estimate the subtraction term
if strcmpi(receiverKnowsCarrier,'True')
    offsetTerm = 2*pi*fc*t+beta; %if carrier frequency & phase offset is known
else
    p = polyfit(t,inst_phase,1); %linearly fit the instaneous phase
    estimated = polyval(p,t); %re-evaluate the offset term using the fitted values
    offsetTerm = estimated;
end
    
demodulated = inst_phase - offsetTerm;

figure()
plot(t,demodulated); %demodulated signal
title('Demodulated signal'); xlabel('n'); ylabel('\hat{m(t)}');

Python code

import numpy as np
from scipy.signal import hilbert
import matplotlib.pyplot as plt
PI = np.pi

fc = 240 #carrier frequency
fm = 10 #frequency of modulating signal
alpha = 1 #amplitude of modulating signal
theta = PI/4 #phase offset of modulating signal
beta = PI/5 #constant carrier phase offset 
receiverKnowsCarrier= False; #If receiver knows the carrier frequency & phase offset

fs = 8*fc #sampling frequency
duration = 0.5 #duration of the signal
t = np.arange(int(fs*duration)) / fs #time base

#Phase Modulation
m_t = alpha*np.sin(2*PI*fm*t + theta) #modulating signal
x = np.cos(2*PI*fc*t + beta + m_t ) #modulated signal

plt.figure()
plt.subplot(2,1,1)
plt.plot(t,m_t) #plot modulating signal
plt.title('Modulating signal')
plt.xlabel('t')
plt.ylabel('m(t)')
plt.subplot(2,1,2)
plt.plot(t,x) #plot modulated signal
plt.title('Modulated signal')
plt.xlabel('t')
plt.ylabel('x(t)')

#Add AWGN noise to the transmitted signal
nMean = 0 #noise mean
nSigma = 0.1 #noise sigma
n = np.random.normal(nMean, nSigma, len(t))
r = x + n  #noisy received signal

#Demodulation of the noisy Phase Modulated signal
z= hilbert(r) #form the analytical signal from the received vector
inst_phase = np.unwrap(np.angle(z))#instaneous phase

#If receiver don't know the carrier, estimate the subtraction term
if receiverKnowsCarrier:
    offsetTerm = 2*PI*fc*t+beta; #if carrier frequency & phase offset is known
else:
    p = np.poly1d(np.polyfit(t,inst_phase,1)) #linearly fit the instaneous phase
    estimated = p(t) #re-evaluate the offset term using the fitted values
    offsetTerm = estimated
                           
demodulated = inst_phase - offsetTerm 

plt.figure()
plt.plot(t,demodulated) #demodulated signal
plt.title('Demodulated signal')
plt.xlabel('n')
plt.ylabel('\hat{m(t)}')

Results

Figure 1: Phase modulation - modulating signal and modulated (transmitted) signal
Figure 1: Phase modulation – modulating signal and modulated (transmitted) signal
Figure 2: Demodulated signal from the noisy received signal

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For further reading

[1] V. Cizek, “Discrete Hilbert transform”, IEEE Transactions on Audio and Electroacoustics, Volume: 18 , Issue: 4 , December 1970.↗

Topics in this chapter

Essentials of Signal Processing
● Generating standard test signals
 □ Sinusoidal signals
 □ Square wave
 □ Rectangular pulse
 □ Gaussian pulse
 □ Chirp signal
● Interpreting FFT results - complex DFT, frequency bins and FFTShift
 □ Real and complex DFT
 □ Fast Fourier Transform (FFT)
 □ Interpreting the FFT results
 □ FFTShift
 □ IFFTShift
● Obtaining magnitude and phase information from FFT
 □ Discrete-time domain representation
 □ Representing the signal in frequency domain using FFT
 □ Reconstructing the time domain signal from the frequency domain samples
● Power spectral density
● Power and energy of a signal
 □ Energy of a signal
 □ Power of a signal
 □ Classification of signals
 □ Computation of power of a signal - simulation and verification
● Polynomials, convolution and Toeplitz matrices
 □ Polynomial functions
 □ Representing single variable polynomial functions
 □ Multiplication of polynomials and linear convolution
 □ Toeplitz matrix and convolution
● Methods to compute convolution
 □ Method 1: Brute-force method
 □ Method 2: Using Toeplitz matrix
 □ Method 3: Using FFT to compute convolution
 □ Miscellaneous methods
● Analytic signal and its applications
 □ Analytic signal and Fourier transform
 □ Extracting instantaneous amplitude, phase, frequency
 □ Phase demodulation using Hilbert transform
● Choosing a filter : FIR or IIR : understanding the design perspective
 □ Design specification
 □ General considerations in design

Books by the author


Wireless Communication Systems in Matlab
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Digital Modulations using Python
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Extract envelope, phase using Hilbert transform: Demo

Key focus: Learn how to use Hilbert transform to extract envelope, instantaneous phase and frequency from a modulated signal. Hands-on demo using Python & Matlab.

If you would like to brush-up the basics on analytic signal and how it related to Hilbert transform, you may visit article: Understanding Analytic Signal and Hilbert Transform

Introduction

The concept of instantaneous amplitude/phase/frequency are fundamental to information communication and appears in many signal processing application. We know that a monochromatic signal of form cannot carry any information. To carry information, the signal need to be modulated. Take for example the case of amplitude modulation, in which a positive real-valued signal modulates a carrier . That is, the amplitude modulation is effected by multiplying the information bearing signal with the carrier signal .

Here, is the angular frequency of the signal measured in radians/sec and is related to the temporal frequency as . The term is also called instantaneous amplitude.

Similarly, in the case of phase or frequency modulations, the concept of instantaneous phase or instantaneous frequency is required for describing the modulated signal.

Here, is the constant amplitude factor (no change in the envelope of the signal) and is the instantaneous phase which varies according to the information. The instantaneous angular frequency is expressed as the derivative of instantaneous phase.

This article is part of the following books
● Digital Modulations using Matlab : Build Simulation Models from Scratch, ISBN: 978-1521493885
● Digital Modulations using Python ISBN: 978-1712321638
● Wireless communication systems in Matlab ISBN: 979-8648350779
All books available in ebook (PDF) and Paperback formats

Definition

Generalizing these concepts, if a signal is expressed as

  • The instantaneous amplitude or the envelope of the signal is given by
  • The instantaneous phase is given by 
  • The instantaneous angular frequency is derived as
  • The instantaneous temporal frequency is derived as

Problem statement

An amplitude modulated signal is formed by multiplying a sinusoidal information and a linear frequency chirp. The information content is expressed as and the linear frequency chirp is made to vary from to . Given the modulated signal, extract the instantaneous amplitude (envelope), instantaneous phase and the instantaneous frequency.

Solution

We note that the modulated signal is a real-valued signal. We also take note of the fact that amplitude/phase and frequency can be easily computed if the signal is expressed in complex form. Which transform should we use such that the we can convert a real signal to the complex plane without altering the required properties ?? Answer: Apply Hilbert transform and form the analytic signal on the complex plane.  Figure 1 illustrates this concept.

Figure 1: Converting a real-valued signal to complex plane using Hilbert Transform

If we express the real-valued modulated signal as an analytic signal, it is expressed in complex plane as

where, (HT[.]) represents the Hilbert Transform operation. Now, the required parameters are very easy to obtain.

The instantaneous amplitude (envelope extraction) is computed in the complex plane as

The instantaneous phase is computed in the complex plane as

The instantaneous temporal frequency is computed in the complex plane as

Once we know the instantaneous phase, the carrier can be regenerated as . The regenerated carrier is often referred as Temporal Fine Structure (TFS) in Acoustic signal processing [1].

Matlab

fs = 600; %sampling frequency in Hz
t = 0:1/fs:1-1/fs; %time base
a_t = 1.0 + 0.7 * sin(2.0*pi*3.0*t) ; %information signal
c_t = chirp(t,20,t(end),80); %chirp carrier
x = a_t .* c_t; %modulated signal

subplot(2,1,1); plot(x);hold on; %plot the modulated signal

z = hilbert(x); %form the analytical signal
inst_amplitude = abs(z); %envelope extraction
inst_phase = unwrap(angle(z));%inst phase
inst_freq = diff(inst_phase)/(2*pi)*fs;%inst frequency

%Regenerate the carrier from the instantaneous phase
regenerated_carrier = cos(inst_phase);

plot(inst_amplitude,'r'); %overlay the extracted envelope
title('Modulated signal and extracted envelope'); xlabel('n'); ylabel('x(t) and |z(t)|');
subplot(2,1,2); plot(cos(inst_phase));
title('Extracted carrier or TFS'); xlabel('n'); ylabel('cos[\omega(t)]');

Python

import numpy as np
from scipy.signal import hilbert, chirp
import matplotlib.pyplot as plt

fs = 600.0 #sampling frequency
duration = 1.0 #duration of the signal
t = np.arange(int(fs*duration)) / fs #time base

a_t =  1.0 + 0.7 * np.sin(2.0*np.pi*3.0*t)#information signal
c_t = chirp(t, 20.0, t[-1], 80) #chirp carrier
x = a_t * c_t #modulated signal

plt.subplot(2,1,1)
plt.plot(x) #plot the modulated signal

z= hilbert(x) #form the analytical signal
inst_amplitude = np.abs(z) #envelope extraction
inst_phase = np.unwrap(np.angle(z))#inst phase
inst_freq = np.diff(inst_phase)/(2*np.pi)*fs #inst frequency

#Regenerate the carrier from the instantaneous phase
regenerated_carrier = np.cos(inst_phase)

plt.plot(inst_amplitude,'r'); #overlay the extracted envelope
plt.title('Modulated signal and extracted envelope')
plt.xlabel('n')
plt.ylabel('x(t) and |z(t)|')
plt.subplot(2,1,2)
plt.plot(regenerated_carrier)
plt.title('Extracted carrier or TFS')
plt.xlabel('n')
plt.ylabel('cos[\omega(t)]')

Results

Figure 2: Amplitude modulation using a chirp signal and extraction of envelope and TFS

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Reference

[1] Moon, Il Joon, and Sung Hwa Hong. “What Is Temporal Fine Structure and Why Is It Important?” Korean Journal of Audiology 18.1 (2014): 1–7. PMC. Web. 24 Apr. 2017.↗

Topics in this chapter

Essentials of Signal Processing
● Generating standard test signals
 □ Sinusoidal signals
 □ Square wave
 □ Rectangular pulse
 □ Gaussian pulse
 □ Chirp signal
● Interpreting FFT results - complex DFT, frequency bins and FFTShift
 □ Real and complex DFT
 □ Fast Fourier Transform (FFT)
 □ Interpreting the FFT results
 □ FFTShift
 □ IFFTShift
● Obtaining magnitude and phase information from FFT
 □ Discrete-time domain representation
 □ Representing the signal in frequency domain using FFT
 □ Reconstructing the time domain signal from the frequency domain samples
● Power spectral density
● Power and energy of a signal
 □ Energy of a signal
 □ Power of a signal
 □ Classification of signals
 □ Computation of power of a signal - simulation and verification
● Polynomials, convolution and Toeplitz matrices
 □ Polynomial functions
 □ Representing single variable polynomial functions
 □ Multiplication of polynomials and linear convolution
 □ Toeplitz matrix and convolution
● Methods to compute convolution
 □ Method 1: Brute-force method
 □ Method 2: Using Toeplitz matrix
 □ Method 3: Using FFT to compute convolution
 □ Miscellaneous methods
● Analytic signal and its applications
 □ Analytic signal and Fourier transform
 □ Extracting instantaneous amplitude, phase, frequency
 □ Phase demodulation using Hilbert transform
● Choosing a filter : FIR or IIR : understanding the design perspective
 □ Design specification
 □ General considerations in design

Books by the author


Wireless Communication Systems in Matlab
Second Edition(PDF)
Note: There is a rating embedded within this post, please visit this post to rate it.
Checkout Added to cart

Digital Modulations using Python
(PDF ebook)
Note: There is a rating embedded within this post, please visit this post to rate it.
Checkout Added to cart

Digital Modulations using Matlab
(PDF ebook)
Note: There is a rating embedded within this post, please visit this post to rate it.
Checkout Added to cart
Hand-picked Best books on Communication Engineering
Best books on Signal Processing

Understanding Analytic Signal and Hilbert Transform

Key focus of this article: Understand the relationship between analytic signal, Hilbert transform and FFT. Hands-on demonstration using Python and Matlab.

Introduction

Fourier Transform of a real-valued signal is complex-symmetric. It implies that the content at negative frequencies are redundant with respect to the positive frequencies. In their works, Gabor [1] and Ville [2], aimed to create an analytic signal by removing redundant negative frequency content resulting from the Fourier transform. The analytic signal is complex-valued but its spectrum will be one-sided (only positive frequencies) that preserved the spectral content of the original real-valued signal. Using an analytic signal instead of the original real-valued signal, has proven to be useful in many signal processing applications. For example, in spectral analysis, use of analytic signal in-lieu of the original real-valued signal mitigates estimation biases and eliminates cross-term artifacts due to negative and positive frequency components [3].

This article is part of the following books
● Digital Modulations using Matlab : Build Simulation Models from Scratch, ISBN: 978-1521493885
● Digital Modulations using Python ISBN: 978-1712321638
● Wireless communication systems in Matlab ISBN: 979-8648350779
All books available in ebook (PDF) and Paperback formats

Continuous-time analytic signal

Let be a real-valued non-bandlimited finite energy signal, for which we wish to construct a corresponding analytic signal .  The Continuous Time Fourier Transform of is given by

Lets say the magnitude spectrum of is as shown in Figure 1(a). We note that the signal is a real-valued and its magnitude spectrum is symmetric and extends infinitely in the frequency domain.

Figure 1: (a) Spectrum of continuous signal x(t) and (b) spectrum of analytic signal z(t)

As mentioned in the introduction, an analytic signal can be formed by suppressing the negative frequency contents of the Fourier Transform of the real-valued signal. That is, in frequency domain, the spectral content of the analytic signal is given by

The corresponding spectrum of the resulting analytic signal is shown in Figure 1(b).

Since the spectrum of the analytic signal is one-sided, the analytic signal will be complex valued in the time domain, hence the analytic signal can be represented in terms of real and imaginary components as . Since the spectral content is preserved in an analytic signal, it turns out that the real part of the analytic signal in time domain is essentially the original real-valued signal itself . Then, what takes place of the imaginary part ? Who is the companion to x(t) that occupies the imaginary part in the resulting analytic signal ? Summarizing as equation,

It is interesting to note that Hilbert transform [4] can be used to find a companion function (imaginary part in the equation above) to a real-valued signal such that the real signal can be analytically extended from the real axis to the upper half of the complex plane . Denoting Hilbert transform as , the analytic signal is given by

From these discussion, we can see that an analytic signal for a real-valued signal , can be constructed using two approaches.

●  Frequency domain approach: The one-sided spectrum of is formed from the two-sided spectrum of the real-valued signal by applying equation (2)
● Time domain approach: Using Hilbert transform approach given in equation (4)

One of the important property of an analytic signal is that its real and imaginary components are orthogonal

Discrete-time analytic signal

Since we are in digital era, we are more interested in discrete-time signal processing. Consider a continuous real-valued signal gets sampled at interval seconds and results in real-valued discrete samples , i.e, . The spectrum of the continuous signal is shown in Figure 2(a). The spectrum of that results from the process of periodic sampling is given in Figure 2(b) (Refer here more details on the process of sampling).  The spectrum of discrete-time signal can be obtained by Discrete-Time Fourier Transform (DTFT).

Figure 2: (a) CTFT of continuous signal x(t), (b) Spectrum of x[n] resulted due to periodic sampling and (c) Periodic one-sided spectrum of analytical signal z[n]

At this point, we would like to construct a discrete-time analytic signal from the real-valued sampled signal . We wish the analytic signal is complex valued and should satisfy the following two desired properties

● The real part of the analytic signal should be same as the original real-valued signal.
● The real and imaginary part of the analytic signal should satisfy the following property of orthogonality

In Frequency domain approach for the continuous-time case, we saw that an analytic signal is constructed  by suppressing the negative frequency components from the spectrum of the real signal. We cannot do this for our periodically sampled signal . Periodic mirroring nature of the spectrum prevents one from suppressing the negative components. If we do so, it will vanish the entire spectrum. One solution to this problem is to set the negative half of each spectral period to zero. The resulting spectrum of the analytic signal is shown in Figure 2(c).

Given a record of samples of even length , the procedure to construct the analytic signal is as follows. This method satisfies both the desired properties listed above.

● Compute the -point DTFT of using FFT
● N-point periodic one-sided analytic signal is computed by the following transform

● Finally, the analytic signal (z[n]) is obtained by taking the inverse DTFT of

Matlab

The given procedure can be coded in Matlab using the FFT function. Given a record of real-valued samples , the corresponding analytic signal can be constructed as given next. Note that the Matlab has an inbuilt function to compute the analytic signal. The in-built function is called hilbert.

function z = analytic_signal(x)
%x is a real-valued record of length N, where N is even %returns the analytic signal z[n]
x = x(:); %serialize
N = length(x);
X = fft(x,N);
z = ifft([X(1); 2*X(2:N/2); X(N/2+1); zeros(N/2-1,1)],N);
end

To test this function, we create a 5 seconds record of a real-valued sine signal. The analytic signal is constructed and the orthogonal components are plotted in Figure 3. From the plot, we can see that the real part of the analytic signal is exactly same as the original signal (which is the cosine signal) and the imaginary part of the analytic signal is phase shifted version of the original signal. We note that the imaginary part of the analytic signal is a cosine function with amplitude scaled by which is none other than the Hilbert transform of sine function.

t=0:0.001:0.5-0.001;
x = sin(2*pi*10*t); %real-valued f = 10 Hz
subplot(2,1,1); plot(t,x);%plot the original signal
title('x[n] - original signal'); xlabel('n'); ylabel('x[n]');

z = analytic_signal(x); %construct analytic signal
subplot(2,1,2); plot(t, real(z), 'k'); hold on;
plot(t, imag(z), 'r');
title('Components of Analytic signal'); 
xlabel('n'); ylabel('z_r[n] and z_i[n]');
legend('Real(z[n])','Imag(z[n])');

Python

Equivalent code in Python is given below (tested with Python 3.6.0)

import numpy as np
def main():
    t = np.arange(start=0,stop=0.5,step=0.001)
    x = np.sin(2*np.pi*10*t)
    
    import matplotlib.pyplot as plt
    plt.subplot(2,1,1)
    plt.plot(t,x)
    plt.title('x[n] - original signal')
    plt.xlabel('n')
    plt.ylabel('x[n]')
    
    z = analytic_signal(x)
    
    plt.subplot(2,1,2)
    plt.plot(t,z.real,'k',label='Real(z[n])')
    plt.plot(t,z.imag,'r',label='Imag(z[n])')
    plt.title('Components of Analytic signal')
    plt.xlabel('n')
    plt.ylabel('z_r[n] and z_i[n]')
    plt.legend()

def analytic_signal(x):
    from scipy.fftpack import fft,ifft
    N = len(x)
    X = fft(x,N)
    h = np.zeros(N)
    h[0] = 1
    h[1:N//2] = 2*np.ones(N//2-1)
    h[N//2] = 1
    Z = X*h
    z = ifft(Z,N)
    return z

if __name__ == '__main__':
    main()
Figure 3: Components of analytic signal for a real-valued sine function

Hilbert Transform using FFT

We should note that the hilbert function in Matlab returns the analytic signal $latex z[n]$ not the hilbert transform of the signal . To get the hilbert transform, we should simply get the imaginary part of the analytic signal. Since we have written our own function to compute the analytic signal, getting the hilbert transform of a real-valued signal goes like this.

x_hilbert = imag(analytic_signal(x))

In the coming posts, we will some of the applications of constructing an analytic signal. For example: Find the instantaneous amplitude and phase of a signal, envelope detector for an amplitude modulated signal, detecting phase changes in a sine wave.

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References:

[1] D. Gabor, “Theory of communications”, Journal of the Inst. Electr. Eng., vol. 93, pt. 111, pp. 42-57, 1946. See definition of complex signal on p. 432.↗
[2] J. A. Ville, “Theorie et application de la notion du signal analytique”, Cables el Transmission, vol. 2, pp. 61-74, 1948.↗
[3] S. M. Kay, “Maximum entropy spectral estimation using the analytical signal”, IEEE transactions on Acoustics, Speech, and Signal Processing, vol. 26, pp. 467-469, October 1978.↗
[4] Frank R. Kschischang, “The Hilbert Transform”, University of Toronto, October 22, 2006.↗
[5] S. L. Marple, “Computing the discrete-time ‘analytic’ signal via FFT,” Conference Record of the Thirty-First Asilomar Conference on Signals, Systems and Computers , Pacific Grove, CA, USA, 1997, pp. 1322-1325 vol.2.↗

Topics in this chapter

Essentials of Signal Processing
● Generating standard test signals
 □ Sinusoidal signals
 □ Square wave
 □ Rectangular pulse
 □ Gaussian pulse
 □ Chirp signal
● Interpreting FFT results - complex DFT, frequency bins and FFTShift
 □ Real and complex DFT
 □ Fast Fourier Transform (FFT)
 □ Interpreting the FFT results
 □ FFTShift
 □ IFFTShift
● Obtaining magnitude and phase information from FFT
 □ Discrete-time domain representation
 □ Representing the signal in frequency domain using FFT
 □ Reconstructing the time domain signal from the frequency domain samples
● Power spectral density
● Power and energy of a signal
 □ Energy of a signal
 □ Power of a signal
 □ Classification of signals
 □ Computation of power of a signal - simulation and verification
● Polynomials, convolution and Toeplitz matrices
 □ Polynomial functions
 □ Representing single variable polynomial functions
 □ Multiplication of polynomials and linear convolution
 □ Toeplitz matrix and convolution
● Methods to compute convolution
 □ Method 1: Brute-force method
 □ Method 2: Using Toeplitz matrix
 □ Method 3: Using FFT to compute convolution
 □ Miscellaneous methods
● Analytic signal and its applications
 □ Analytic signal and Fourier transform
 □ Extracting instantaneous amplitude, phase, frequency
 □ Phase demodulation using Hilbert transform
● Choosing a filter : FIR or IIR : understanding the design perspective
 □ Design specification
 □ General considerations in design

Books by the author


Wireless Communication Systems in Matlab
Second Edition(PDF)
Note: There is a rating embedded within this post, please visit this post to rate it.
Checkout Added to cart

Digital Modulations using Python
(PDF ebook)
Note: There is a rating embedded within this post, please visit this post to rate it.
Checkout Added to cart

Digital Modulations using Matlab
(PDF ebook)
Note: There is a rating embedded within this post, please visit this post to rate it.
Checkout Added to cart
Hand-picked Best books on Communication Engineering
Best books on Signal Processing