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

Use of a bilinear conformal map to achieve a frequency warping nearly identical to that of the Bark frequency scale is described Because the map takes the unit circle to itself, its form is that of the transfer function of a first-order allpass filter Since it is a first-order map, it preserves the model order of rational systems, making it a valuable frequency warping technique for use in audio filter design A closed-form weighted-equation-error method is derived that computes the optimal mapping coefficient as a function of sampling rate, and the solution is shown to be generally indistinguishable from the optimal least-squares solution The optimal Chebyshev mapping is also found to be essentially identical to the optimal least-squares solution The expression 08517[arctan(006583fs)]/sup 1/2/-0916 is shown to accurately approximate the optimal allpass coefficient as a function of sampling rate f/sub s/ in kHz for sampling rates greater than 1 kHz A filter design example is included that illustrates improvements due to carrying out the design over a Bark scale Corresponding results are also given and compared for approximating the related "equivalent rectangular bandwidth (ERB) scale" of Moore and Glasberg (ACTA Acustica, vo82, p335-45, 1996) using a first-order allpass transformation Due to the higher frequency resolution called for by the ERB scale, particularly at low frequencies, the first-order conformal map is less able to follow the desired mapping, and the error is two to three times greater than the Bark-scale case, depending on the sampling rate

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Bark and ERB bilinear transforms

The proliferation of MPEG coded audio material on the Internet has shown an exponential growth since 1995, making ”.mp3” the most searched for term in early 1999 (according to http://www.searchterms.com). ”MP3” has been featured in numerous articles in newspapers and periodicals and on TV, mostly on the business pages because of the potential impact on the recording industry. While everybody is using MP3, not many (including some of the software authors writing MP3 encoders, decoders or associated tools) know the history and the details of MPEG audio coding. This paper explains the basic technology and some of the special features of MPEG1/2 Layer-3 (aka MP3). It also sheds some light on the factors determining the quality of compressed audio and what can be done wrong in MPEG encoding and decoding.

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MP3 and AAC Explained

Spectral Band Replication, a Novel Approach in Audio Coding

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 ISO/IEC MPEG-2 advanced audio coding (AAC) system was designed to provide MPEG-2 with the best audio quality without any restrictions due to compatibility requirements. The main features of the AAC system (ISO/IEC 13818-7) are described. MPEG-2 AAC combines the coding efficiency of a high-resolution filter bank, prediction techniques, and Huffman coding with additional functionalities aimed to deliver very high audio quality at a variety of data rates.

ISO/IEC MPEG-2 Advanced Audio Coding

The invention relates to a method for coding or de-coding an audio signal combining the advantages of TNS processing and noise substitution. A time discrete audio signal is initially transformed in a frequency range in order to obtain spectral value of the temporal audio signal. A prediction of the spectral values in relation to frequency is subsequently made in order to enable spectral residual values. Areas within the spectral values encompassing spectral values with noise properties are detected . The spectral residual values are noise substituted in the noise areas, whereupon data relating to the noise areas and noise substitution are incorporated into side information pertaining to a coded audio signal.

Method for coding an audio signal

In the coding and decoding of stereo audio spectral values both the intensity stereo process and prediction are used in order to achieve high data compression. If intensity stereo coding is active in one section of scale factor bands, the prediction for the right channel in that range is deactivated, whereby the results of the prediction are not used to form the coded stereo audio spectral values. To allow further adaptation of the prediction for the right channel, the predictor of the right channel is fed with stereo audio spectral values for the channel, which again are intensity stereo decoded.

Coding and decoding of audio signals by using intensity stereo and prediction processes

A method of encoding time-discrete audio signals comprises the steps of weighting the time-discrete audio signal by means of window functions overlapping each other so as to form blocks, the window functions producing blocks of a first length for signals varying weakly with time and blocks of a second length for signals varying strongly with time. A start window sequence is selected for the transition from windowing with blocks of the first length to windowing with blocks of the second length, whereas a stop window sequence is selected for the opposite transition. The start window sequence is selected from at least two different start window sequences having different lengths, whereas the stop window sequence is selected from at least two different stop window sequences having different lengths. A method of decoding blocks of encoded audio signals selects a suitable inverse transformation as well as a suitable synthesis window as a reaction to side information associated with each block.

Martin Dietz

Papers

ISO/IEC MPEG-2 Advanced Audio Coding

Spectral Band Replication, a Novel Approach in Audio Coding

Method for coding an audio signal

Coding and decoding of audio signals by using intensity stereo and prediction processes

Method subband of coding and decoding audio signals using variable length windows