During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low cost. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linear prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby AC-2/AC-3, and the Sony ATRAC/SDDS algorithms. Then, we describe subjective evaluation methodologies in some detail, including the ITU-R BS.1116 recommendation on subjective measurements of small impairments. This paper concludes with a discussion of future research directions.

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Perceptual coding of digital audio

Foundations of modern auditory theory

With the advent of `multimedia', digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing to become a research field of its own. To date, most research in DSP applied to sound has been concentrated on speech, which is bandwidth limited to about 4 kilohertz. Speech processing is also limited by the low fidelity typically expected in the telephone network. Today, the main applications of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including perceptual measurement techniques and analysis/synthesis methods. Additional important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application-oriented research. The frequency range of wideband audio has an upper limit of 20 kilohertz and the resulting difference in frequency range and Signal to Noise Ratio (SNR) due to sample size must be taken into account when designing DSP algorithms. There are whole classes of algorithms that the speech community is not interested in pursuing or using. These algorithms and techniques are revealed in this book. This book is suitable for advanced level courses and serves as a valuable reference for researchers in the field. Interested and informed engineers will also find the book useful in their work.

Applications of digital signal processing to audio and acoustics

The invention relates in general to the recording, transmitting, and reproducing of multidimensional sound fields intended for human hearing. More particularly, the invention relates to subband encoding and decoding of signals representing such sound fields, wherein the encoded signals may be carried by multiple discrete signals and/or a composite signal with a control signal conveying either the relative levels of the encoded signals, or the apparent direction of the sound field represented by the encoded signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signals.

Encoder/decoder for multidimensional sound fields

Describes a novel wavelet based audio synthesis and coding method. The method uses optimal adaptive wavelet selection and wavelet coefficients quantization procedures together with a dynamic dictionary approach. The adaptive wavelet transform selection and transform coefficient bit allocation procedures are designed to take advantage of the masking effect in human hearing. They minimize the number of bits required to represent each frame of audio material at a fixed distortion level. The dynamic dictionary greatly reduces statistical redundancies in the audio source. Experiments indicate that the proposed adaptive wavelet selection procedure by itself can achieve almost transparent coding of monophonic compact disk (CD) quality signals (sampled at 44.1 kHz) at bit rates of 64-70 kilobits per second (kb/s). The combined adaptive wavelet selection and dynamic dictionary coding procedures achieve almost transparent coding of monophonic CD quality signals at bit rates of 48-66 kb/s. >

Low bit rate transparent audio compression using adapted wavelets

The exploitation of left-right correlation in a subband code for stereophonic audio signals is investigated. A transform of left and right signals into decorrelated intensity and error signals is presented. Although this can be seen as the optimal exploitation of redundancy, it yields only marginal gain in bit rate. If the reduced phase-sensitivity of the human observer can be exploited by encoding only the intensity signal, a substantial gain can be obtained. Preliminary results of a stereo codec are promising: at 192 kb/s good coding results have been obtained. >

Subband coding of stereophonic digital audio signals

Subband coding of digital audio signals

A subband coding system for high quality digital audio signals is described. To achieve low bit rates at a high quality level, it exploits the simultaneous masking effect of the human ear. It is shown how this effect can be used in an adaptive bit-allocation scheme. The proposed approach has been applied in two coding systems, a complex system in which signal is split into 26 subbands, each approximately one third of an octave wide, and a simpler 20-band system. Both systems have been designed for coding stereophonic 16-bit compact disk signals with a sampling frequency of 44.1 kHz. With the 26-band system high-quality results can be obtained at bit rates of 220 kb/s. With the 20-band system, similar results can be obtained at bit rates of 360 kb/s. >

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Subband coding of digital audio signals without loss of quality

Since 1988 ISO/IEC JTCI/SC29 WG11 (MPEG) is working on the standardization of video and audio signals. The Audio subgroup of MPEG is working on bit rate reduction systems for high quality digital audio. Since the first phase of this standardization effort has been finished, MPEG/Audio is extending its work to multichannel audio coding systems as well as to medium quality coding at lower sampling frequencies and lower bit rates. Future standardization work aims at next-generation coder suitable for high quality audio transmission and storage at bit rates of 64 kb/s per channel and well below. >

http://ieeexplore.ieee.org/iel2/3039/8632/00380000.pdf

R.G. van der Waal

Papers

Subband coding of stereophonic digital audio signals

Subband coding of digital audio signals

Subband coding of digital audio signals without loss of quality

Current and future standardization of high-quality digital audio coding in MPEG