Analysis of the MPEG-1 Layer III (MP3) Algorithm using MATLAB

by Andreas Spanias, Jayaraman Thiagarajan

Overview

The MPEG-1 Layer III (MP3) algorithm is one of the most successful audio formats for consumer audio storage and for transfer and playback of music on digital audio players. The MP3 compression standard along with the AAC (Advanced Audio Coding) algorithm are associated with the most successful music players of the last decade. This book describes the fundamentals and the MATLAB implementation details of the MP3 algorithm. Several of the tedious processes in MP3 are supported by demonstrations using MATLAB software. The book presents the theoretical concepts and algorithms used in the MP3 standard. The implementation details and simulations with MATLAB complement the theoretical principles. The extensive list of references enables the reader to perform a more detailed study on specific aspects of the algorithm and gain exposure to advancements in perceptual coding. Table of Contents: Introduction / Analysis Subband Filter Bank / Psychoacoustic Model II / MDCT / Bit Allocation, Quantization and Coding / Decoder

Product Details

ISBN-13:	9783031003905
Publisher:	Springer International Publishing
Publication date:	12/01/2011
Series:	Synthesis Lectures on Algorithms and Software in Engineering
Pages:	115
Product dimensions:	7.52(w) x 9.25(h) x (d)

About the Author

Jayaraman J.Thiagarajan completed his MS in Electrical Engineering and is currently a PhD candidate in the School of Electrical, Computer, and Energy Engineering at Arizona State University. His research interests are in the areas of DSP, audio processing and coding, sparse representations and computer vision. His works in sparse representations have been nominated for awards at IEEE DSP Workshop 2011 and Asilomar 2011. He has worked extensively with the MPEG-1 Layer-III audio coding standard and has created MATLAB modules for the various functions of the algorithm. In addition, he has developed LabVIEW interfaces for teaching speech coding algorithms and contributed to the Java-DSP software package. Andreas Spanias is a Professor in the School of Electrical, Computer, and Energy Engineering at Arizona State University (ASU). He is also the founder and director of the SenSIP Center and NSF NCSS I/UCRC site. His research interests are in the areas of adaptive signal processing, speech processing, and audio sensing. He and his student team developed the computer simulation software Java-DSP ( JDSP– ISBN 0-9724984-0-0). He is author of two text books: Audio Processing and Coding by Wiley and DSP: An Interactive Approach. He served as Associate Editor of the IEEE Transactions on Signal Processing and as General Co-chair of IEEE ICASSP-99. He also served as the IEEE Signal Processing Vice-President for Conferences. Andreas Spanias is co-recipient of the 2002 IEEE Donald G. Fink paper prize award and was elected Fellow of the IEEE in 2003. He served as Distinguished lecturer for the IEEE Signal processing society in 2004.

Table of Contents

Introduction.- Analysis Subband Filter Bank.- Psychoacoustic Model II.- MDCT.- Bit Allocation, Quantization and Coding.- Decoder.