This paper presents two fundamental enhancements in a hybrid audio signal model consisting of sinusoidal, transient, and noise (STN) components. The first enhancement involves a novel application of a perceptual metric for optimal time segmentation for the analysis of transients. In particular, Moore and Glasberg's model of partial loudness is modified for use with general signals and then integrated into a novel time segmentation scheme. The second, and perhaps more significant STN enhancement is concerned with a new methodology for ranking and selection of the most perceptually relevant sinusoids. A systematic procedure is developed for the selection of a compact set of sinusoids and comparative results are given to demonstrate the merit of this method.

This paper presents two fundamental enhancements in a hybrid audio signal model consisting of sinusoidal, transient, and noise (STN) components. The first enhancement involves a novel application of a perceptual metric for optimal time segmentation for the analysis of transients. In particular, Moore and Glasberg's model of partial loudness is modified for use with general signals and then integrated into a, novel time segmentation scheme. The second and perhaps more significant STN enhancement is concerned with a new methodology for ranking and selection of the most perceptually relevant sinusoids.

This paper presents a new method for selecting the sinusoids in a hybrid audio signal model consisting of sinusoidal, transient, and noise (STN) components. The method relies on a new methodology for ranking and selection of the most perceptually relevant sinusoids. We call this method excitation similarity weighting (ESW) ranking and selection procedure. Whereas current coders tend to choose maximum signal-to-mask ratio components the ESW methodology seeks to maximize the matching between the excitation patterns evoked by the coded and original signals on a short-time basis. It is shown that the proposed method provides a perceptual improvement.

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.

A GUI-based software tool that provides a framework for evaluation of different speech coding algorithms is presented. The tool is designed to measure the susceptibility of speech coding algorithms to errors added on the encoded bit-stream during transmission. In particular, the errors can be added individually to the parameters that comprise the encoded bit-stream. This enables a designer of a speech codec to evaluate its performance under adverse or impaired channel conditions. The tool is universally applicable to different speech coding algorithms, by means of a user-defined bit-stream definition file. In fact, the tool has been used in the past to evaluate a number of standardized speech coding algorithms. This paper describes the features of the software. In addition, the paper presents sample results generated by this tool during a study of the ETSI GSM Enhanced Full Rate algorithm.

A MATLAB educational simulation program was developed to explore and understand standardized speech coding algorithms such as the FS-1015 LPC-10e, the FS-1016 CELP, the ETSI GSM, the IS-54 VSELP, the G.721 ADPCM, and the G.728 LD-CELP algorithms. The simulation software provides an interactive environment that allows users to display time- and frequency-domain representations of input and reconstructed speech on an IBM PC compatible equipped with a standard sound card. Scores associated with subjective and objective performance measures are computed using simulation tools. This tool is used in undergraduate and graduate DSP courses to introduce speech coding fundamentals.