6 edition of Digital signal processing found in the catalog.
Includes bibliographical references (p. 1053-1066) and index.
|Other titles||Digital signal processing :|
|Statement||John G. Proakis, Dimitris G. Manolakis.|
|Contributions||Manolakis, Dimitris G.|
|LC Classifications||TK5102.9 .P757 2007|
|The Physical Object|
|Pagination||xix, 1084 p. :|
|Number of Pages||1084|
As such, this title is also useful to undergraduates in electrical engineering, and as a reference for science students and practicing engineers. Contents and Highlights Chapters 1 and 2 contain a discussion of the two key DSP concepts of sampling and quantization. Practicing engineers and graduate students may also find it useful as a first text on the subject. In comparison, fricative sounds originate from the noisy air turbulence at narrow constrictions, such as the teeth and lips. It explains the principal approaches together with the most recent mathematical models which are being used worldwide.
The second part is more advanced and discusses the practical issues of choosing and defining specifications for antialiasing prefilters and anti-image postfilters. In the block processing part, we discuss convolution and several ways Digital signal processing book thinking about it, transient and steady-state behavior, and real-time processing on a block-by-block basis using the overlap-add method and its software implementation. A filter may also be described as a difference equationa collection of zeros and poles or an impulse response or step response. Where phase is unimportant, often the Fourier transform is converted to the power spectrum, which is the magnitude of each frequency component squared.
Each word in the incoming audio signal is isolated and then analyzed to identify the type of excitation and resonate frequencies. With some applications, how the phase varies with frequency can be a significant consideration. Filtering, particularly in non-realtime work can also be achieved in the frequency domain, applying the filter and then converting back to the time domain. The engineer can study the spectrum to determine which frequencies are present in the input signal and which are missing.
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This is the most common method of digital speech generation used today. Digital filtering generally consists of some linear transformation of a number of surrounding samples around the current Digital signal processing book of the input or output signal.
Personal information is secured with SSL technology. The included computer experiments can form the basis Digital signal processing book an accompanying DSP lab course, as is done at Rutgers.
The first part emphasizes the issues of spectral analysis, frequency resolution, windowing, and leakage. It is then low-pass filtered and downscaled, yielding an approximation image; this image is high-pass filtered to produce the three smaller detail images, and low-pass filtered to produce the final approximation image in the upper-left.
Weitzen, whose comments helped improve the book. Two approaches are used for computer generated speech: digital recording and vocal tract simulation. This emphasizes the harmonic structure of the original spectrum. Digital Signal Processing generally approaches the problem of voice recognition in two steps: feature extraction followed by feature matching.
Time and space domains[ edit ] Main article: Time domain The most common processing approach in the time or space domain is enhancement of the input signal through a method called filtering. As such, this title is also useful to undergraduates in electrical engineering, and as a reference for science students and practicing engineers.
Quantization effects are briefly discussed. In digital recording, the voice of a human speaker is digitized and stored, usually in a compressed form. We present several practical design examples of interpolation filters, including polyphase and multistage designs.
It can be covered quickly as most of this material is assumed known from a prerequisite linear systems course. With voiced sounds, vocal cord vibration produces near periodic pulses of air into the vocal Digital signal processing book.
Linear filters satisfy the superposition principlei. The tradeoff between oversampling ratio and savings in bits is derived. We present a systematic treatment of the subject and carry it on to the remainder of the book.
Main article: Sampling signal processing To digitally analyze and manipulate an analog signal, it must be digitized Digital signal processing book an analog-to-digital converter ADC. Numerical methods require a quantized signal, such as those produced by an ADC.
The standard model of quantization noise is presented, as well as the techniques of oversampling, noise shaping, and dithering. Signal Sampling and Quantization Objectives 2. Sound originates in the vocal tract in one of two basic ways, called voiced and fricative sounds. An entire hour of recorded speech requires only about three megabytes of storage, well within the capabilities of even small computer systems.
Michell Publisher: IN-TECH January 16, The book aims to guide the students, engineers, researchers and the industrial community on the latest updates in the design and implementation of DSP systems and its utility in programmable devices and application specific circuits.
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It is ebook good book, lots of theory in there, however it is a bit dry in its explanations, there are a lot of results (for example the value of a certain number series) that just are presented out of the blue while the reader scratches his head trying to come up with the missing steps/5(8).