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Journal of the Audio Engineering Society

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2006 December - Volume 54 Number 12

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Papers

Equalization in an Extended Area Using Multichannel Inversion and Wave Field Synthesis

Authors: Corteel, Etienne

Wave field synthesis (WFS) targets the synthesis of the physical characteristics of a sound field in an extended listening area. This synthesis is, however, accompanied by noticeable reconstruction artifacts. They are due to both loudspeaker radiation characteristics and approximations to the underlying physical principles. These artifacts may introduce coloration, which must be compensated for over the entire listening area. Multichannel equalization techniques allow for the control of the sound field produced by a loudspeaker array at a limited number of positions. The control can be extended to a large portion of space by employing a new method that combines multichannel equalization with a linear microphone array–based description of the sound field and accounts for WFS rendering characteristics and limitations. The proposed method is evaluated using an objective coloration criterion. Its benefits compared to conventional equalization techniques are pointed out for both ideal omnidirectional loudspeakers and multi-actuator panels.

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Filter Design Method for Loudspeaker Equalization Based on IIR Parametric Filters

Authors: Ramos, Germán; López, José J.

A novel method for the equalization of loudspeakers and other audio systems using IIR (infinite impulse response) parametric filters is presented. The main characteristic of the proposed filter design method resides in the fact that the equalization structure is planned from the beginning as a chain of SOSs (second-order sections), where each SOS is a low-pass, high-pass, or peak filter, defined by its parameters. The algorithm combines a direct search method with a heuristic parametric optimization process where constraints on the values could be imposed in order to obtain practical implementations. A psychoacoustic model based on the detection of peaks and dips in the frequency response has been employed to determine which ones need to be equalized, reducing the filter order without noticeable effect. The first computed sections of the designed filter are the ones that correct the response more effectively, allowing scalable solutions when hardware limitations exist or different degrees of correction are needed. The method has been validated with subjective testing and compared with other methods. Its results could be applied to passive and multiway active loudspeakers.

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Perceptually Biased Linear Prediction

Authors: Biswas, Arijit; den Brinker, Albertus C.

A perceptually biased linear prediction scheme is proposed for audio coding. Using only simple modifications of the coefficients defining the normal equations for a least-squares error, the spectral masking effects are mimicked in the prediction synthesis filter without using an explicit psychoacoustic model. The main advantage of such a scheme is reduced computational complexity. The proposed approach was implemented in a Laguerre-based linear prediction scheme, and its performance was evaluated in comparison with a Laguerrebased linear prediction approach controlled by the ISO MPEG-1 Layer I–II model, as well as with one of the latest spectral integration–based psychoacoustic models. Listening tests clearly demonstrate the viability of the proposed method.

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Application and Verfication of the Objective Quality Assessment Method According to ITU Recommendation Series ITU-T P.862

Authors: Kurittu, Antti; Samela, Juha; Lakaniemi, Ari; Mattila, Ville-veikko; Zacharov, Nick

[Engineering Report] Listening speech quality has for many years been considered a key means of evaluating speech quality in telecommunications systems. While traditionally listening tests have been used to study such characteristics, such as in the form of mean opinion scores (MOS), these methods are known to be quite cumbersome and slow. As a result a number of speech quality prediction algorithms have been developed and standardized over the last few years, allowing for estimation of the subjective speech quality based upon the physical characteristics of the speech signal. The most commonly used and standardized predictive algorithms are considered, namely, ITU-T recommendations P.862/P.862.1 and P.862.2. The prediction performance of these algorithms is evaluated under common testing conditions for both narrow- and wide-band speech signals. Applicability in the context of spectral aberration, time scaling, noise, and environmental noise is considered as well as the prediction of codec performance.

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Forging Ahead with Spatial Audio Coding

Authors: Staff, AES

[Feature Article] Spatial audio coding is a term recently coined to describe low bit-rate audio coding that enables the transmission of multichannel spatial audio at fairly low bit rates. In a typical spatial audio coder (see Fig. 1) the audio signal is downmixed to a relatively small number of basic channels (usually one or two), and transmitted along with additional information that enables the reconstruction of the missing channels. Such “parametric” coding includes the calculation of interchannel level, time/phase, and correlation, which are the key parameters of multichannel audio signals containing information about the spatial characteristics of the overall scene. The bit rates employed can be very low compared with traditional multichannel audio codecs, with spatial side chain bit rates of between 32 and 48 kbit/s, in addition to whatever bit rate is employed for the primary channels. The primary channel is often coded using a legacy system for compatibility of the downmixed audio with existing decoders.

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