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Journal of the Acoustical Society of America

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Jan 2010

Volume 127, Issue 1, pp. EL1-611

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Design of an efficient music-speech discriminator

Lorenzo J. Tardón, Simone Sammartino, and Isabel Barbancho

J. Acoust. Soc. Am. Volume 127, Issue 1, pp. 271-279 (2010); (9 pages)

Online Publication Date: 05 Jan 2010

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In this paper, the problem of the design of a simple and efficient music-speech discriminator for large audio data sets in which advanced music playing techniques are taught and voice and music are intrinsically interleaved is addressed. In the process, a number of features used in speech-music discrimination are defined and evaluated over the available data set. Specifically, the data set contains pieces of classical music played with different and unspecified instruments (or even lyrics) and the voice of a teacher (a top music performer) or even the overlapped voice of the translator and other persons. After an initial test of the performance of the features implemented, a selection process is started, which takes into account the type of classifier selected beforehand, to achieve good discrimination performance and computational efficiency, as shown in the experiments. The discrimination application has been defined and tested on a large data set supplied by Fundación Albéniz, containing a large variety of classical music pieces played with different instrument, which include comments and speeches of famous performers.
Show PACS
43.60.Dh Signal processing for communications: telephony and telemetry, sound pickup and reproduction, multimedia
43.60.Lq Acoustic imaging, displays, pattern recognition, feature extraction
43.75.Xz Automatic music recognition, classification, and information retrieval

Intelligibility and listener preference of telephone speech in the presence of babble noise

Joseph L. Hall and James L. Flanagan

J. Acoust. Soc. Am. Volume 127, Issue 1, pp. 280-285 (2010); (6 pages) | Cited 2 times

Online Publication Date: 05 Jan 2010

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Two procedures for improving the intelligibility of wideband telephone speech in the presence of competing babble noise are evaluated. One procedure is differentiation, the other consists of equalizing the speech spectrum by applying the inverse of the average spectrum of formant amplitudes for adult male speakers (“formant equalization”). Speech processed by these two methods was evaluated both for intelligibility and for listener preference. Both methods produced a clear increase in intelligibility compared to unprocessed wideband telephone speech. Formant equalization was found to be preferred over differentiation, more so at low signal-to-noise ratios than at high ones.
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43.60.Dh Signal processing for communications: telephony and telemetry, sound pickup and reproduction, multimedia
43.66.Dc Masking
43.66.Lj Perceptual effects of sound
43.71.Gv Measures of speech perception (intelligibility and quality)

A different approach to use narrowband super-resolution multiple signal classification algorithm on wideband sources

Mohammad Asgari, Nasim Yahya Soltani, and Ali Riahi

J. Acoust. Soc. Am. Volume 127, Issue 1, pp. 286-291 (2010); (6 pages)

Online Publication Date: 05 Jan 2010

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There are varieties of wideband direction-of-arrival (DOA) estimation algorithms. Their structure comprises a number of narrowband ones, each performs in one frequency in a given bandwidth, and then different responses should be combined in a proper way to yield true DOAs. Hence, wideband algorithms are always complex and so non-real-time. This paper investigates a method to derive a flat response of narrowband multiple signal classification (MUSIC) [ R. O. Schmidt, IEEE Trans. Antennas Propag., 34, 276–280 (1986) ] algorithm in the whole frequencies of given band. Therefore, required conditions of applying narrowband algorithm on wideband impinging signals will be given through a concrete analysis. It could be found out that array sensor locations are able to compensate the frequency variations to reach a flat response of DOAs in a specified wideband frequency.
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43.60.Fg Acoustic array systems and processing, beam-forming

Distribution theory approach to implementing directional acoustic sensors

Dean J. Schmidlin

J. Acoust. Soc. Am. Volume 127, Issue 1, pp. 292-299 (2010); (8 pages)

Online Publication Date: 05 Jan 2010

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The objective of directional acoustic sensors is to provide high directivity while occupying a small amount of space. An idealized point sensor achieves this objective from a knowledge of the spatial partial derivatives of acoustic pressure at a point in space. Direct measurement of these derivatives is difficult in practice. Consequently, it is expedient to come up with indirect methods. The use of pressure sensors to construct finite-difference approximations is an example of such a method. This paper utilizes the theory of distributions to derive another indirect method for estimating the various spatial partial derivatives of the pressure. This alternate method is then used to construct a multichannel filter which processes the acoustic pressure by mean of three-dimensional integral transforms throughout a 6ε-length cube centered at the origin. The output of the multichannel filter is a spatially and temporally filtered version of the pressure at the origin. The temporal filter is a lowpass Gaussian filter whose bandwidth is inversely proportional to ε. Finally, the lattice method for numerical multiple integration is utilized to develop a discrete-spatial version of the multichannel filter.
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43.60.Fg Acoustic array systems and processing, beam-forming
43.30.Wi Passive sonar systems and algorithms, matched field processing in underwater acoustics

Reconstruction of radiating sound fields using minimum energy method

Rolf Bader

J. Acoust. Soc. Am. Volume 127, Issue 1, pp. 300-308 (2010); (9 pages) | Cited 1 time

Online Publication Date: 05 Jan 2010

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A method for reconstructing a pressure field at the surface of a radiating body or source is presented using recording data of a microphone array. The radiation is assumed to consist of many spherical radiators, as microphone positions are present in the array. These monopoles are weighted using a parameter α, which broadens or narrows the overall radiation directivity as an effective and highly intuitive parameter of the radiation characteristics. A radiation matrix is built out of these weighted monopole radiators, and for different assumed values of α, a linear equation solver reconstructs the pressure field at the body’s surface. It appears that from these many arbitrary reconstructions, the correct one minimizes the reconstruction energy. The method is tested, localizing the radiation points of a Balinese suling flute, reconstructing complex radiation from a duff frame drum, and determining the radiation directivity for the first seven modes of an Usbek tambourine. Stability in terms of measurement noise is demonstrated for the plain method, and additional highly effective algorithm is added for a noise level up to 0 dB. The stability of α in terms of minimal reconstruction energy is shown over the whole range of possible values for α. Additionally, the treatment of unwanted room reflections is discussed, still leading to satisfactory results in many cases.
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43.60.Fg Acoustic array systems and processing, beam-forming
43.60.Gk Space-time signal processing, other than matched field processing
43.75.Kk Bells, gongs, cymbals, mallet percussion, and similar instruments

Short pulse multi-frequency phase-based time delay estimation

Said Assous, Clare Hopper, Mike Lovell, Dave Gunn, Peter Jackson, and John Rees

J. Acoust. Soc. Am. Volume 127, Issue 1, pp. 309-315 (2010); (7 pages) | Cited 2 times

Online Publication Date: 05 Jan 2010

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An approach for time delay estimation, based on phase difference detection, is presented. A multiple-frequency short continuous wave pulse is used to solve the well-known phase ambiguity problem when the maximum distance exceeds a full wavelength. Within an unambiguous range defined with the lowest frequency difference between components, the corresponding phase difference is unique and any distance within this range can be determined. Phase differences between higher frequency components are used to achieve a finer resolution. The concept will be presented and the effectiveness of the approach will be investigated through theoretical and practical examples. The method will be validated using underwater acoustic measurements, simulating noisy environments, demonstrating resolutions better than a 50th of a wavelength, even in the presence of high levels (−5 dB) of additive Gaussian noise. Furthermore, the algorithm is simple to use and can be easily implemented, being based on phase detection using the discrete Fourier transform.
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43.60.Jn Source localization and parameter estimation
43.60.Ac Theory of acoustic signal processing
43.60.Bf Acoustic signal detection and classification, applications to control systems
43.60.Qv Signal processing instrumentation, integrated systems, smart transducers, devices and architectures, displays and interfaces for acoustic systems
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