A simulation study about the CT technique employing a stationary sound wave with long wavelength is presented. The CT is an inverse and nonlinear problem, so that an iterative approach is inevitable. That is, starting an assumed density distribution, the field calculation is repeatedly made until a true density distribution is achieved. An inhomogeneous domain is divided into subregions in which the density is assumed to be constant. The Finite Element Method is used for calculating the acoustic field under the given particle velocity driving source and the Newton-Raphson method is used to update the initial density distribution by minimizing the scalar error between computed and measured field values. To cope with the ill-condition which essentially exist in the inverse problem, three kinds of methods are used comparatively to solve this problem. For getting a more exact reconstruction, a new algorithm employing prior information is proposed. The simulation results show that the proposed algorithm can reconstruct satisfactorily the density distribution image of the model by the proper selection of driving position and its frequency.
In this study, we investigated the effect of pitch frequency in forming the attentional filters of complex tone and frequency-gliding tones in signal detection tasks. The attentional filters were measured at spectral regions where there is no real power or no power convergence on a single frequency. In Experiment I, the attentional filter around the missing-fundamental frequency was measured by the probe-signal method. The cue tone was a complex tone composed of 13 components from 1,000 Hz up to 4,000 Hz, whose fundamental frequency is 250 Hz. In Experiment II, we also investigated the formation of attentional filters in relation to frequency-gliding tonal cues. The frequency was changed from 925 Hz to 1,075 Hz in an upward-frequency glide and from 1,075 Hz to 925 Hz in a downward-frequency glide. The overall pitch was first measured by pitch-matching and the attentional filter was then measured around the overall pitch frequency. In conclusion, an attentional filter can be formed at the fundamental frequency region, where is no real power, and at the frequency corresponding to overall pitch of frequency-gliding tone.
In this paper, we present a VLSI implementation method on a dynamic range controller, audio level compressor, which is used to make sound effects as an important functional part in a digital audio system on a chip. To implement the gain calculation in a digital dynamic range controller, a power calculation with fractional numbers is required and it is difficult to be performed directly in a digital audio system. We introduce a polynomial expression to approximate the power operation, then the gain calculation is easily performed with a number of additions, multiplications and a division. Based on the gain calculation method of the gain, an efficient VLSI architecture with some arithmetic circuits is proposed. We designed the circuit of a 16-bit audio level compressor by using a hardware description language, VHDL. As a result of the compact circuit design, the 16-bit compressor has only 5,688 gates by 1 μm CMOS gate array technology. The design and simulation results show that the presented audio level compressor has high performance and can be integrated into an audio system on a chip for practical use.
A physiological articulatory model has been developed to simulate the dynamic actions of speech organs during speech production. This model represents the midsagittal region of the tongue, jaw, hyoid bone, and the vocal tract wall in three dimensions. The soft tissue of the tongue is outlined in the midsagittal and parasagittal planes of MR images obtained from a male Japanese speaker, and constructed as a 2-cm thick layer. The palatal and pharyngeal walls are constructed as a hard shell of a 3-cm left-to-right width. The jaw and hyoid bone are modelled to yield rotation and translation motions. The muscle structure in the model is identified based on volumetric MR images of the same speaker. A fast simulation method is developed by modeling both the soft tissue and rigid organs using mass-points with two types of links: viscoelastic springs with a proper stiffness for connective tissue, and extremely high stiffness for bony organs. Muscle activation signals are generated by a model control strategy based on the target-reaching task, and then fed to drive the model to approach the targets. The model demonstrated realistic behaviors similar to coarticulation in human speech production (Dang and Honda, 1998, 1999, 2000).
An IIR implementation of the gammachirp filter has been proposed to simulate basilar membrane motion efficiently (Irino and Unoki, 1999). A reasonable filter response was provided by a combination of a gammatone filter and an IIR asymmetric compensation (AC) filter. It was noted, probably however, that the rms error was high when the absolute values of the parameters are large, because the coefficients of the IIR-AC filter were selected heuristically. In this report, we show that this is due to the sign inversion of the phase of poles and zeros in the conventional model. We propose a new definition of the IIR-AC filter and we describe a method of systematic determining the optimum coefficients and number of cascade for the second-order filter. This results in a reduction of the error to about 1/3 that produced by the conventional model.
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