John Wiley, 2008. — 334 p.Digital audio signal processing is employed in recording and storing music and speech signals, for sound mixing and production of digital programs, in digital transmission to broadcast receivers as well as in consumer products like CDs, DATs and PCs. In the latter case, the audio signal is in a digital form all the way from the microphone right up to the loudspeakers, enabling real-time processing with fast digital signal processors.
This book provides the basis of an advanced course in Digital Audio Signal Processing which I have been giving since 1992 at the Technical University Hamburg-Harburg. It is directed at students studying engineering, computer science and physics and also for professionals looking for solutions to problems in audio signal processing like in the fields of studio engineering, consumer electronics and multimedia. The mathematical and theoretical fundamentals of digital audio signal processing systems will be presented and typical applications with an emphasis on realization aspects will be discussed. Prior knowledge of systems theory, digital signal processing and multirate signal processing is taken as a prerequisite.
The book is divided into two parts. The first part (Chapters 1–4) presents a basis for hardware systems used in digital audio signal processing. The second part (Chapters 5–9) discusses algorithms for processing digital audio signals. Chapter 1 describes the course taken by an audio signal from its recording in a studio up to its reproduction at home. Chapter 2 contains a representation of signal quantization, dither techniques and spectral shaping of quantization errors used for reducing the nonlinear effects of quantization. In the end, a comparison is made between the fixed-point and floatingpoint number representations as well as their associated effects on format conversion and algorithms. Chapter 3 describes methods for AD/DA conversion of signals, starting with Nyquist sampling, methods for oversampling techniques and delta-sigma modulation. The chapter closes with a presentation of some circuit design of AD/DA converters. After an introduction to digital signal processors and digital audio interfaces, Chapter 4 describes simple hardware systems based on a single- and multiprocessor solutions. The algorithms introduced in the following Chapters 5–9 are, to a great extent, implemented in real-time on hardware platforms presented in Chapter 4_. Chapter 5 describes digital audio equalizers. Apart from the implementation aspects of recursive audio filters, nonrecursive linear phase filters based on fast convolution and filter banks are introduced. Filter designs, parametric filter structures and precautions for reducing quantization errors in recursive filters are dealt with in detail. Chapter 6 deals with room simulation. Methods for simulation of artificial room impulse response and methods for approximation of measured impulse responses are discussed. In Chapter 7 the dynamic range control of audio signals is described. These methods are applied at several positions in the audio chain from the microphone up to the loudspeakers in order to adapt to the dynamics of the recording, transmission and listening environment. Chapter 8 contains a presentation of methods for synchronous and asynchronous sampling rate conversion. Efficient algorithms are described which are suitable for real-time processing as well as off-line processing. Both lossless and lossy audio coding are discussed in Chapter 9_. Lossless audio coding is applied for storing of higher word-lengths. Lossy audio coding, on the other hand, plays a significant role in communication systems.Introduction
Quantization
AD/DA Conversion
Audio Processing Systems
Equalizers
Room Simulation
Dynamic Range Control
Sampling Rate Conversion
Audio Coding
