Introduction To Signals And Systems And Digital Signal Processing

If you understand basic mathematics and know how to program with Python, you’re ready to dive into signal processing. While most resources start with theory to teach this complex subject, this practical book introduces techniques by showing you how they’re applied in the real world. In the first chapter alone, you’ll be able to decompose a sound into its harmonics, modify the harmonics, and generate new sounds. Author Allen Downey explains techniques such as spectral decomposition, filtering, convolution, and the Fast Fourier Transform. This book also provides exercises and code examples to help you understand the material. You’ll explore: Periodic signals and their spectrums Harmonic structure of simple waveforms Chirps and other sounds whose spectrum changes over time Noise signals and natural sources of noise The autocorrelation function for estimating pitch The discrete cosine transform (DCT) for compression The Fast Fourier Transform for spectral analysis Relating operations in time to filters in the frequency domain Linear time-invariant (LTI) system theory Amplitude modulation (AM) used in radio Other books in this series include Think Stats and Think Bayes, also by Allen Downey.

Introduction to Digital Signal Processing covers the basic theory and practice of digital signal processing (DSP) at an introductory level. As with all volumes in the Essential Electronics Series, this book retains the unique formula of minimal mathematics and straightforward explanations. The author has included examples throughout of the standard software design package, MATLAB and screen dumps are used widely throughout to illustrate the text. Ideal for students on degree and diploma level courses in electric and electronic engineering, 'Introduction to Digital Signal Processing' contains numerous worked examples throughout as well as further problems with solutions to enable students to work both independently and in conjunction with their course. Assumes only minimum knowledge of mathematics and electronics Concise and written in a straightforward and accessible style Packed with worked examples, exercises and self-assesment questions

Signals, Systems, Transforms, and Digital Signal Processing with MATLAB® has as its principal objective simplification without compromise of rigor. Graphics, called by the author, "the language of scientists and engineers", physical interpretation of subtle mathematical concepts, and a gradual transition from basic to more advanced topics are meant to be among the important contributions of this book. After illustrating the analysis of a function through a step-by-step addition of harmonics, the book deals with Fourier and Laplace transforms. It then covers discrete time signals and systems, the z-transform, continuous- and discrete-time filters, active and passive filters, lattice filters, and continuous- and discrete-time state space models. The author goes on to discuss the Fourier transform of sequences, the discrete Fourier transform, and the fast Fourier transform, followed by Fourier-, Laplace, and z-related transforms, including Walsh–Hadamard, generalized Walsh, Hilbert, discrete cosine, Hartley, Hankel, Mellin, fractional Fourier, and wavelet. He also surveys the architecture and design of digital signal processors, computer architecture, logic design of sequential circuits, and random signals. He concludes with simplifying and demystifying the vital subject of distribution theory. Drawing on much of the author’s own research work, this book expands the domains of existence of the most important transforms and thus opens the door to a new world of applications using novel, powerful mathematical tools.

This text provides a basic understanding of digital signal processing concepts and techniques. It begins with the characterization of discrete-time signals and systems in the time and frequency domains augmented by MATLAB functions. It then covers Fourier analysis based on digital techniques.

"An excellent introductory book" (Review of the First Edition in the International Journal of Electrical Engineering Education) " it will serve as a reference book in this area for a long time" (Review of Revised Edition in Zentralblatt für Mathematik (Germany)) Firmly established as the essential introductory Digital Signal Processing (DSP) text, this second edition reflects the growing importance of random digital signals and random DSP in the undergraduate syllabus by including two new chapters. The authors' practical, problem-solving approach to DSP continues in this new material, which is backed up by additional worked examples and computer programs. The book now features: * fundamentals of digital signals and systems * time and frequency domain analysis and processing, including digital convolution and the Discrete and Fast Fourier Transforms * design and practical application of digital filters * description and processing of random signals, including correlation, filtering, and the detection of signals in noise Programs in C and equivalent PASCAL are listed in an Appendix. Typical results and graphic plots from all the programs are illustrated and discussed in the main text. The overall approach assumes no prior knowledge of electronics, computing, or DSP. An ideal text for undergraduate students in electrical, electronic and other branches of engineering, computer science, applied mathematics and physics. Practising engineers and scientists will also find this a highly accessible introduction to an increasingly important field.

With an interesting approach to educate the students in signals and systems, and digital signal processing simultaneously, this book not only provides a comprehensive introduction to the basic concepts of the subject but also offers a practical treatment of the modern concepts of digital signal processing. Written in a cogent and lucid manner, the book is addressed to the needs of undergraduate engineering students of electrical, electronics, and computer disciplines, for a first course in signals and digital signal processing.

"The DFT can be understood as a numerical approximation to the Fourier transform. However, the DFT has its own exact Fourier theory, and that is the focus of this book. The DFT is normally encountered as the Fast Fourier Transform (FFT)--a high-speed algorithm for computing the DFT. The FFT is used extensively in a wide range of digital signal processing applications, including spectrum analysis, high-speed convolution (linear filtering), filter banks, signal detection and estimation, system identification, audio compression (such as MPEG-II AAC), spectral modeling sound synthesis, and many others. In this book, certain topics in digital audio signal processing are introduced as example applications of the DFT"--Back cover

1. Señales y sistemas 2. Sistemas lineales invariantes en el tiempo 3. Representación de señales periódicas en series de Fourier 4. La transformada contínua de Fourier 5. La transformada de Fourier de tiempo discreto 6. Caracterización en tiempo y frecuencia de señales y sistemas 7. Muestreo 8. Sistemas de comunicación 9. La transformada de Laplace 10. La transformada z 11. Sistemas lineales retroalimentados.