Showing papers by "Andreas Spanias published in 2000"

Perceptual coding of digital audio

Abstract: 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.

Development and evaluation of a Web-based signal and speech processing laboratory for distance learning

Abstract: We describe an Internet-based signal processing laboratory that provides hands-on learning experiences in distributed learning environments. The laboratory is based on an object-oriented Java/sup TM/ tool called Java Digital Signal Processing (J-DSP). J-DSP has been developed at Arizona State University (ASU) and is being used for a virtual laboratory in a senior-level DSP course. J-DSP is written as a platform-independent Java applet that resides on the Web and is thereby accessible by all students through the use of a Web browser. J-DSP has a rich suite of signal processing functions that facilitate interactive on-line simulations of modern statistical signal and spectral analysis algorithms, filter design tools, QMF banks, and state-of-the-art vocoders. J-DSP is accompanied by administrative software tools for secure Internet-based lab-report submission and evaluation including servlets for maintaining Web-based grade books. A series of J-DSP laboratory exercises has been developed and delivered using the ASU distance learning facilities. Student evaluations as well as assessments by experts have been compiled and preliminary results are quite encouraging.

Minimum-variance phase prediction and frame interpolation algorithms for low bit rate sinusoidal speech coding

Abstract: A number of improved algorithms for phase prediction and frame interpolation in the context of sinusoidal speech coding are presented. A minimum-variance sinusoidal phase estimation scheme is proposed. It is shown that reasonably accurate estimates for short-time sinusoidal phases corresponding to voiced frames can be obtained. In addition, improved algorithms for interpolation of sine wave parameters are presented which result in further reduction in bit rate while preserving the subjective equality of the reproduced speech at low bit rates. The performance of the proposed algorithms were evaluated on a large speech database and the results of statistical analysis are provided. The proposed algorithms were successfully integrated into a 2.4 kbps sinusoidal coder, where speech of good quality intelligibility, and naturalness was obtained.