A MUSICAM source codec for digital audio broadcasting and storage

Dehery, Yves-Francois; Lever, M.; Urcun, P.

doi:10.1109/icassp.1991.151054

Cited by 30 publications

(7 citation statements)

References 4 publications

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“…Based primarily upon coders developed at IRT and Phillips, the MUSICAM algorithm [10][117] was successful in the 1990 ISO/IEC competition [118] for a new audio coding standard. It eventually formed the basis for MPEG-1 and MPEG-2 audio layers I and II.…”

Section: B Musicammentioning

confidence: 99%

“…Much of this work was motivated by standardization activities for the European Eureka-147 digital broadcast audio (DBA) system. The ISO/IEC eventually clustered the IRT, Philips, and CCETT proposals into a single candidate algorithm, Masking Pattern Adapted Universal Subband Integrated Coding and Multiplexing (MUSICAM) [10][117], which competed successfully for inclusion in the ISO/IEC MPEG-1 and MPEG-2 audio coding standards. Consequently, most of MPEG-1 [17] and MPEG-2 [18] layers I and II are derived from MUSICAM.…”

Section: Subband Codersmentioning

confidence: 99%

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Perceptual coding of digital audio

2000

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I. INTRODUCTIONAudio coding or audio compression algorithms are used to obtain compact digital representations of high-fidelity (wideband) audio signals for the purpose of efficient transmission or storage. The central objective in audio coding is to represent the signal with a minimum number of bits while achieving transparent signal reproduction, i.e., generating output audio that cannot be distinguished from the original input, even by a sensitive listener ("golden ears"). This paper gives a review of algorithms for transparent coding of high-fidelity audio.The introduction of the compact disk (CD) in the early eighties [1] brought to the fore all of the advantages of digital audio representation, including unprecedented high-fidelity, dynamic range, and robustness. These advantages, however, came at the expense of high data rates. Conventional CD and digital audio tape (DAT) systems are typically sampled at either 44.1 or 48 kilohertz (kHz) using pulse code modulation (PCM) with a sixteen bit sample resolution. This results in uncompressed data rates of 705.6/768 kilobits per second (kbps) for a monaural channel, or 1.41/1.54 megabits per second (Mbps) for a stereo pair at 44.1/48 kHz, respectively. Although high, these data rates were accommodated successfully in first generation digital audio applications such as CD and DAT. Unfortunately, second generation multimedia applications and wireless systems in particular are often subject to bandwidth and cost constraints that are incompatible with high data rates. Because of the success enjoyed by the first generation, however, end users have come to expect "CD-quality" audio reproduction from any digital system. Therefore, new network and wireless multimedia digital audio systems must reduce data rates without compromising reproduction quality. These and other considerations have motivated considerable research during the last decade towards formulation of compression schemes that can satisfy simultaneously the conflicting demands of high compression ratios and transparent reproduction quality for high-fidelity audio signals

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Section: B Musicammentioning

confidence: 99%

Section: Subband Codersmentioning

confidence: 99%

Perceptual coding of digital audio

2000

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“…Although transform coding is widely used in image, video, and audio coding, for example, JPEG, MPEG video, and MPEG audio [8][9][10][11][12], its application in speech coding is far from successful. If speech coding is used for wireless communication, transform coding in its elementary form may not be suitable.…”

Section: Internetmentioning

confidence: 99%

A Simple MDCT-Based Speech Coder for Internet Applications

Kwong

Pang

et al. 1999

Lecture Notes in Computer Science

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Abstract.A very simple speech coder is investigated based on modified discrete-cosine transform (MDCT), run-length coding, and entropy code. The speech signal is first passed through a MDCT, quantized by a uniform quantizer, and run-length coded to achieve good compression ratio. Not suitable for wireless communication systems, the speech coder is perfect for Internet and speech storage applications. With a performance far better than most speech coders for wireless applications, low data-rate speech of 3 kbit/s can be achieved with a signal-to-noise ratio (S/N) of about 12 dB.

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“…An estimation of the tonelike or noiselike quality for Cj is then obtained LISing the spectral flatness measure [40] (SFM) (10) where p A and ,U(! correspond to the geometric and arithmetic means of the PSD components for each band.…”

Section: Temporal Maskingmentioning

confidence: 99%

A review of algorithms for perceptual coding of digital audio signals

Painter

Spanias

Proceedings of 13th International Conference on Digital Signal Processing

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During the last decade, CD-quality digital audio has essentially replaced analog audio. During this same period, new digital audio applications have emerged for network, wireless, and multimedia computing systems which face such constraints as reduced channel bandwidth, limited storage capacity, and low cost. These new applications have creuted(1 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 algorithmsfor 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 commercicil product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. The paper is organized as follows. First, psychoacoustic principles are described with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Then, we review methodologies which achieve perceptually transparent coding of FM-and CD-quality audio signals, including algorithms which manipulate transform components and subband signal decompositions. The discussion concentrates on architectures and applications of irhose techniques which utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms which have become international andlor commercial standards are also presented, including the ISO/MPEG family and the Dolby AC-3 algorithms. The paper concludes with a brief discussion of future research divect i o m

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A MUSICAM source codec for digital audio broadcasting and storage

Cited by 30 publications

References 4 publications

Perceptual coding of digital audio

Perceptual coding of digital audio

A Simple MDCT-Based Speech Coder for Internet Applications

A review of algorithms for perceptual coding of digital audio signals

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