Felipe Orduña‐Bustamante

Fast deconvolution of multichannel systems using regularization

A very fast deconvolution method, which is based on the fast Fourier transform (FFT), can be used to control the outputs from a multichannel plant comprising any number of control sources and error sensors. The result is a matrix of causal finite impulse response filters whose performance is optimized at a large number of discrete frequencies. The paper is particularly aimed at multichannel sound reproduction and more specifically reproducing the sound field from a set of loudspeakers.

Inverse filter design and equalization zones in multichannel sound reproduction

A discussion is given of two techniques for designing inverse filters for use in multichannel sound reproduction systems. The first is the multiple-input/output inverse filtering theorem, which uses direct inversion of a matrix containing the coefficients of filters used to specify the electroacoustic transmission paths. The second is an adaptive technique based on the multiple error LMS algorithm. The theory presented reconciles the two approaches and furthermore, derives explicit conditions which must be fulfilled if an exact inverse is to exist. A formula is derived which gives the number of coefficients required in the inverse filters in terms of the number of coefficients used to represent the transmission paths. Some numerical examples are also presented which illustrate the dependence of the mean square error on both the choice of modeling delay and the number of coefficients in the inverse filters. Finally, the results of some simulations are given which demonstrate the acoustical possibilities associated with these filtering techniques. >

Local sound field reproduction using digital signal processing

This work shows how an acoustic wavefront can be reconstructed locally by using only a few loudspeakers. The loudspeaker inputs are calculated by passing a set of signals recorded by only a few microphones through a matrix of causal digital filters having finite impulse responses. These filters, referred to as the inverse filters, are calculated by inverting (in the least‐squares sense) a matrix which contains the electroacoustic transfer functions from the loudspeakers to the microphones. In practice, it is crucial to use a modeling delay and a regularization factor in order to achieve an accurate inversion. The technique is illustrated with an example that shows how well four loudspeakers can reproduce a sound field that has been recorded with three microphones. When the recorded field does not contain energy at frequencies whose acoustical wavelengths are shorter than the distance between adjacent microphones, the original field is reproduced remarkably accurately in the vicinity of the microphones reg...

Nonuniform beams with harmonically related overtones for use in percussion instruments

Beams used in certain percussion instruments with definite pitch such as marimba, xylophone, and vibraphone are undercut in order to bring the frequencies of the first few overtones into a harmonic relationship with the fundamental frequency. This paper addresses the problem of determining the optimal dimensions of the undercut so that the frequencies of modes 2 and 3 of transverse motion are harmonically related with that of the fundamental (mode 1). The undercut is assumed to have a parabolic shape described by two variable parameters related to its depth and width. The rest of the beam on each side of the undercut is assumed to be uniform. Seven pairs of optimal undercut parameters are found that bring the frequencies of the first three transverse modes into the harmonic (integer) ratios 1:3:6, 1:4:8–9, and 1:5:10–13. Calculations performed with and without taking into account the effects of rotary inertia and shear stress are compared against measurements taken from a set of experimental beams. The co...

Nonlinear progressive waves in a slide trombone resonator

The propagation of finite-amplitude waves inside a slide trombone is studied through direct pressure measurements corresponding to dynamic extremes. A two-microphone method is used to separate left-moving and right-moving waves inside the trombone, permitting the detection of nonlinear effects associated with progressive waves. It is found that a redistribution of energy across the spectrum toward the higher-frequencies occurs for large distances and high initial pressure levels. These results are consistent with the theory of weakly nonlinear acoustics and also with those reported in this same context by other authors, but which have been obtained mostly through examination of standing-waves.

Subjective and physical experiments related to the tuning of classical guitars.

The subject of tuning the classical guitar is discussed in the light of subjective experiments during which guitarists tuned a classical guitar using different tuning methods. Tones of the six open strings were recorded after each tuning session, and the perceived musical pitch was estimated using a psychoacoustically based pitch detection algorithm. Also, the input mechanical mobility was measured at the top plate bridge, at the nut bridge, and at all the frets. Frequency shifts due to string‐body coupling were estimated for the frequency components of all possible (open and fretted) guitar tones, and perceived musical pitch was also estimated for these tones. Results are presented of tuning accuracy and variability for individual guitarists as well as among different guitarists. Also, estimates are presented of overall guitar tuning accuracy, depending on string‐body coupling, fretted‐tone over‐tension, tuning method, and other factors.

Practical determination of acoustic parameters of the singing voice implemented in the interactive analysis software EVOCANTO.

Practical methods of analysis with low computational complexity are presented for the determination of acoustic parameters from radiated sound pressure spectra of sustained sung vowels. Sound spectra are analyzed in order to obtain the fundamental vocal source frequency F0 and its harmonics, the corresponding musical pitch, the first few “formant” frequencies F1, F2, possibly up to F5, that are apparent from the harmonic contour, a nominal vocal tract shape derived from (F1, F2), a measure of formant‐to‐harmonic tuning of F1 and F2, indication of the presence of a singer’s formant, and a measure of the rate of vibrato. Real‐time determination and display of these parameters are integrated into the singing voice analysis software EVOCANTO, a graphical interactive didactic computer program which is intended as a teaching aid for singing tutors and students. Examples of the use of EVOCANTO are presented for typical male and female singing voice registers.

Subjective evaluation of classical guitars

An analysis is presented of two methods for the subjective evaluation of a set of classical guitars. One is based on ‘‘blind’’ listening tests by a general audience, and another is based on direct inspection and playing by guitar teachers and students. Subjective experiments reveal that the two types of tests produce essentially the same sorting of the guitars by subjective quality. The listening test, based on pairwise comparisons of guitars playing the same musical fragment, is judged by the subjects to be a difficult task. This is also reflected in the results. Additionally, an interesting effect is consistently observed in which a higher proportion of votes is obtained by the second guitar played in each pair, despite the fact that listening to each pair of guitars lasts less than about 40 s. This seems to suggest a certain natural limit in the short‐term memory abilities of the subjects in recalling the first guitar. A data processing technique is described that produces unbiased results from this ty...

Use of measured data in the physical modeling of the classical guitar

A simple physical model consisting of a delay‐network representation of the string and a radiation filter (which transduces bridge motion into sound pressure radiated at a distance) is used to synthesize the sound of the classical guitar. Parameters for the sound generation algorithm are obtained from mechanical and acoustical transfer functions measured in a guitar. These correspond to mechanical response at the bridge, or to acoustical response at a given point in the sound field, and involve mechanical excitation in the bridge at different string positions. The variation from string to string observed in these measurements, especially at high frequencies, suggests the need for individual characterization of each string in the physical model of the guitar. Other factors, such as stiffness (inharmonicity), internal string damping, and nonlinear string motion are also discussed as desirable additional features of more realistic sound generation algorithms.

Modified acoustic transmission tube apparatus incorporating an active downstream termination

Current techniques for measuring normal incidence sound transmission loss with a modified impedance tube, or transmission tube, require setting up two different absorbing termination loads at the end of the downstream tube [ASTM E2611-09, Standard Test Method for Measurement of Normal Incidence Sound Transmission of Acoustical Materials Based on the Transfer Matrix Method (American Society for Testing and Materials, West Conshohocken, 2009)]. The process of physically handling the two required passive absorbing loads is a possible source of measurement errors, which are mainly due to changes in sample test position, or in test setup re-assembly, between measurements. In this paper, a modified transmission tube apparatus is proposed for non-intrusively changing the downstream acoustic load by means of a combined passive-active termination. It provides a controlled variable sound absorption which simplifies the setup of standard two-load techniques, without the need of physically handling the apparatus during the tests. This virtually eliminates the risk of errors associated with the physical manipulation of the two passive terminations. Transmission loss measurements in some representative test conditions are reported, showing improvements over current implementations, in reducing by approximately 50% the measurement variations associated with the setup of the two required absorbing terminations. Measurement results agree within 0.4 dB (maximum difference in high resolution broadband), and 0.04 dB (mean difference in 1/3-octave bands), with those obtained using standard passive two-load methods.