Digital Signal Processing (DSP) From Ground Up™ in C
Digital Signal Processing (DSP) From Ground Up™ in C, available at $59.99, has an average rating of 4.14, with 92 lectures, based on 758 reviews, and has 5345 subscribers.
You will learn about Be able to develop the Convolution Kernel algorithm in C Be able able to develop the Discrete Fourier Transform (DFT) algorithm in C Be able to develop the Inverse Discrete Fourier Transform (IDFT) algorithm in C Be able to develop the Fast Fourier Transform (FFT) algorithm in C Be able to perform spectral analysis on ECG signals in C Be able to design and develop Windowed-Sinc filters in C Be able to design and develop Finite Impulse Response (FIR) filters in C Be able to design and develop Infinite Impulse Response (IIR) filters in C Be able to develop the FFT Convolution algorithm in C Be able to develop the First Difference algorithm in C Be able to develop the Running Sum algorithm in C Be able to develop the Moving Average filter algorithm in C Be able to develop the Recursive Moving Average filter algorithm in C Be able to develop signal statistical algorithms in C Be able to build passive Low-pass and High-pass filters Be able to build Modified Sallen-Key filters Be able to build Bessel, Chebyshev and Butterworth filters Understand all about Linear Systems and their characteristics Understand how to synthesize and decompose signals Understand the relationship between the delta function and the Impulse response Be able to plot signals with gnuplot Be able to give a lecture on Digital Signal Processing (DSP) Be able to suppress noise in signals This course is ideal for individuals who are Engineering Students or C Developers or Embedded Systems Engineers or Computer Engineering students or Hobbyists or Embedded Systems Instructors It is particularly useful for Engineering Students or C Developers or Embedded Systems Engineers or Computer Engineering students or Hobbyists or Embedded Systems Instructors.
Enroll now: Digital Signal Processing (DSP) From Ground Up™ in C
Summary
Title: Digital Signal Processing (DSP) From Ground Up™ in C
Price: $59.99
Average Rating: 4.14
Number of Lectures: 92
Number of Published Lectures: 92
Number of Curriculum Items: 92
Number of Published Curriculum Objects: 92
Original Price: $119.99
Quality Status: approved
Status: Live
What You Will Learn
- Be able to develop the Convolution Kernel algorithm in C
- Be able able to develop the Discrete Fourier Transform (DFT) algorithm in C
- Be able to develop the Inverse Discrete Fourier Transform (IDFT) algorithm in C
- Be able to develop the Fast Fourier Transform (FFT) algorithm in C
- Be able to perform spectral analysis on ECG signals in C
- Be able to design and develop Windowed-Sinc filters in C
- Be able to design and develop Finite Impulse Response (FIR) filters in C
- Be able to design and develop Infinite Impulse Response (IIR) filters in C
- Be able to develop the FFT Convolution algorithm in C
- Be able to develop the First Difference algorithm in C
- Be able to develop the Running Sum algorithm in C
- Be able to develop the Moving Average filter algorithm in C
- Be able to develop the Recursive Moving Average filter algorithm in C
- Be able to develop signal statistical algorithms in C
- Be able to build passive Low-pass and High-pass filters
- Be able to build Modified Sallen-Key filters
- Be able to build Bessel, Chebyshev and Butterworth filters
- Understand all about Linear Systems and their characteristics
- Understand how to synthesize and decompose signals
- Understand the relationship between the delta function and the Impulse response
- Be able to plot signals with gnuplot
- Be able to give a lecture on Digital Signal Processing (DSP)
- Be able to suppress noise in signals
Who Should Attend
- Engineering Students
- C Developers
- Embedded Systems Engineers
- Computer Engineering students
- Hobbyists
- Embedded Systems Instructors
Target Audiences
- Engineering Students
- C Developers
- Embedded Systems Engineers
- Computer Engineering students
- Hobbyists
- Embedded Systems Instructors
With a programming based approach, this course is designed to give you a solid foundation in the most useful aspects of Digital Signal Processing (DSP) in an engaging and easy to follow way. The goal of this course is to present practical techniques while avoiding obstacles of abstract mathematical theories. To achieve this goal, the DSP techniques are explained in plain language, not simply proven to be true through mathematical derivations.
Still keeping it simple, this course comes in different programming languages and hardware architectures so that students can put the techniques to practice using a programming language or hardware architecture of their choice. This version of the course uses the C programming language.
By the end of this course you should be able develop the Convolution Kernelalgorithm in C, develop the Discrete Fourier Transform (DFT) algorithm in C, develop the Inverse Discrete Fourier Transform (IDFT) algorithm in C, design and develop Finite Impulse Response (FIR) filters in C, design and develop Infinite Impulse Response (IIR) filters in C, develop Windowed-Sinc filters in C, build Modified Sallen-Key filters, build Bessel, Chebyshevand Butterworthfilters, develop the Fast Fourier Transform (FFT) algorithm in C , even give a lecture on DSP and so much more. Please take a look at the full course curriculum.
Course Curriculum
Chapter 1: Set up
Lecture 1: Setting up an Integrated Development Environment (IDE)
Lecture 2: Overview of CodeBlocks
Lecture 3: Downloading gnuplot
Lecture 4: Installing gnuplot
Lecture 5: Overview of gnuplot
Chapter 2: Getting started with gnuplot
Lecture 1: Plotting signals with gnuplot
Lecture 2: Plotting multiple signals in the same window
Chapter 3: Signal Statistics and Noise
Lecture 1: Signal Statistics and Noise
Lecture 2: Mean and Standard Deviation
Lecture 3: Coding : Developing the Signal Mean algorithm
Lecture 4: Coding : Computing the Signal Mean
Lecture 5: Coding : Developing the Signal Variance algorithm
Lecture 6: Coding : Developing the Signal Standard Deviation algorithm
Chapter 4: Quantization and The Sampling Theorem
Lecture 1: Nyquist Theorem ( Sampling Theorem )
Lecture 2: The Passive Low-Pass Filter
Lecture 3: The Passive High-Pass Filter
Lecture 4: The Active Filter
Lecture 5: The Bessel, Chebyshev and Butterworth filters
Chapter 5: Linear Systems and Superposition
Lecture 1: Notice
Lecture 2: Introduction to Linear Systems
Lecture 3: Understanding Superposition
Lecture 4: Impulse and Step Decomposition
Chapter 6: Convolution
Lecture 1: Introduction to Convolution
Lecture 2: The Convolution Operation
Lecture 3: Examinging the Output of Convolution
Lecture 4: The Convolution Sum Equation
Lecture 5: A Closer look at the Delta function
Lecture 6: Coding : Developing the Convolution algorithm (Part I )
Lecture 7: Coding : Developing the Convolution algorithm (Part I I)
Lecture 8: Coding : Developing the Convolution algorithm (Part III)
Lecture 9: Coding : Developing the Convolution algorithm (Part IV)
Lecture 10: The Running Sum and First Difference
Lecture 11: Coding : Developing the Running Sum algorithm
Chapter 7: Fourier Transsform
Lecture 1: Introduction to Fourier Analysis
Lecture 2: The DFT Engine
Lecture 3: Understanding Forward and Inverse DFT
Lecture 4: Code : Developing the DFT algorithm (Part I)
Lecture 5: Code : Developing the DFT algorithm (Part II)
Lecture 6: Code : Developing the DFT algorithm (Part III)
Lecture 7: Coding : Developing the Inverse DFT algorithm (Part I)
Lecture 8: Coding : Developing the Inverse DFT algorithm (Part II)
Lecture 9: Coding : Developing the Inverse DFT algorithm (Part III)
Lecture 10: Coding : Computing the DFT and IDFT of an ECG signal (Part I)
Lecture 11: Coding : Computing the DFT and IDFT of an ECG signal (Part II)
Lecture 12: Coding : Identifying the frequencies present in the DFT plot
Lecture 13: Symmetry between Time domain and frequency domain -Duality
Lecture 14: Polar Notation
Lecture 15: Coding : Rectangular notation to the polar notation ( Part I)
Lecture 16: Coding : Rectangular notation to the polar notation ( Part II)
Chapter 8: Complex Numbers
Lecture 1: The Complex Number System
Lecture 2: Polar Representation of Complex Numbers
Lecture 3: Euler's Relation
Lecture 4: Representation of Sinusoids
Lecture 5: Representing Systems
Chapter 9: Complex Fourier Transform
Lecture 1: Introduction to Complex Fourier Transform
Lecture 2: Mathematical Equivalence
Lecture 3: The Complex DFT Equation
Lecture 4: Comparing Real DFT and Complex DFT
Lecture 5: Coding : Developing the Complex DFT equation (Part I)
Lecture 6: Coding : Developing the Complex DFT equation (Part II )
Chapter 10: Fast Fourier Transform (FFT)
Lecture 1: An Overview of how FFT works.
Lecture 2: Understanding the complexity of calculating DFT directly
Lecture 3: How the Decimation -in-Time FFT Algorithm works
Chapter 11: Digital Filter Design
Lecture 1: Introduction to Digital Filters
Lecture 2: The Filter Kernel
Lecture 3: The Impulse,Step and Frequency response
Lecture 4: Understanding the Logarithmic scale and decibels
Lecture 5: Information representations of a signal
Lecture 6: Time domain parameters
Lecture 7: Frequency domain parameters
Lecture 8: Designing digital filters using the spectral inversion method
Lecture 9: Designing digital filters using the spectral reversal method
Lecture 10: Classification of digital filters
Chapter 12: Designing Finite Impulse Response FIR) Filters
Lecture 1: The Moving Average Filter
Lecture 2: The Multiple Pass Moving Average Filter
Lecture 3: The Recursive Moving Average Filter
Chapter 13: Designing Infinite Impulse Response (IIR) Filters
Lecture 1: Introduction to Recursive Filters
Lecture 2: The Recursion Equation
Lecture 3: The Single-Pole Recursive Filter
Lecture 4: Digital Chebyshev Filters
Chapter 14: Designing Windowed-Sinc Filters
Lecture 1: Introduction to Windowed-Sinc Filters
Lecture 2: The Sinc Function and the Truncated Sinc Filter
Lecture 3: The Blackman window
Lecture 4: The Hamming and Blackman window equations
Lecture 5: Designing the Windowed Sinc filter
Lecture 6: Coding : Developing the Low-pass Windowed-Sinc Filter Algorithm (Part I)
Instructors
-
Israel Gbati
Embedded Firmware Engineer -
BHM Engineering Academy
21st Century Engineering Academy
Rating Distribution
- 1 stars: 15 votes
- 2 stars: 39 votes
- 3 stars: 141 votes
- 4 stars: 236 votes
- 5 stars: 327 votes
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