scholarly journals Localization in a Head-Related Transfer Function-based virtual audio synthesis using additional high-pass and low-pass filtering of sound sources

2007 ◽  
Vol 28 (4) ◽  
pp. 244-250 ◽  
Author(s):  
György Wersényi
2013 ◽  
Vol 109 (6) ◽  
pp. 1658-1668 ◽  
Author(s):  
Daniel J. Tollin ◽  
Janet L. Ruhland ◽  
Tom C. T. Yin

Sound localization along the azimuthal dimension depends on interaural time and level disparities, whereas localization in elevation depends on broadband power spectra resulting from the filtering properties of the head and pinnae. We trained cats with their heads unrestrained, using operant conditioning to indicate the apparent locations of sounds via gaze shift. Targets consisted of broadband (BB), high-pass (HP), or low-pass (LP) noise, tones from 0.5 to 14 kHz, and 1/6 octave narrow-band (NB) noise with center frequencies ranging from 6 to 16 kHz. For each sound type, localization performance was summarized by the slope of the regression relating actual gaze shift to desired gaze shift. Overall localization accuracy for BB noise was comparable in azimuth and in elevation but was markedly better in azimuth than in elevation for sounds with limited spectra. Gaze shifts to targets in azimuth were most accurate to BB, less accurate for HP, LP, and NB sounds, and considerably less accurate for tones. In elevation, cats were most accurate in localizing BB, somewhat less accurate to HP, and less yet to LP noise (although still with slopes ∼0.60), but they localized NB noise much worse and were unable to localize tones. Deterioration of localization as bandwidth narrows is consistent with the hypothesis that spectral information is critical for sound localization in elevation. For NB noise or tones in elevation, unlike humans, most cats did not have unique responses at different frequencies, and some appeared to respond with a “default” location at all frequencies.


2018 ◽  
Vol 11 (2) ◽  
pp. 56-59 ◽  
Author(s):  
Olimpiu Pop ◽  
Corneliu Rusu ◽  
Lacrimioara Grama

Abstract The aim of this work is to develop a device capable to record multiple audio signals (in our case 4 audio signals from 4 microphones of the area) and transmit the information through a network for acoustic source localization. We briefly discuss the first two versions, then the HRTF (Head Related Transfer Function) version of the acoustic sensors is detailed. Experimental results for identifying sound sources are also presented.


2021 ◽  
Vol 24 (3) ◽  
pp. 689-714
Author(s):  
David Kubanek ◽  
Jaroslav Koton ◽  
Jan Jerabek ◽  
Darius Andriukaitis

Abstract The formula of the all-pole low-pass frequency filter transfer function of the fractional order (N + α) designated for implementation by non-cascade multiple-feedback analogue structures is presented. The aim is to determine the coefficients of this transfer function and its possible variants depending on the filter order and the distribution of the fractional-order terms in the transfer function. Optimization algorithm is used to approximate the target Butterworth low-pass magnitude response, whereas the approximation errors are evaluated. The interpolated equations for computing the transfer function coefficients are provided. An example of the transformation of the fractional-order low-pass to the high-pass filter is also presented. The results are verified by simulation of multiple-feedback filter with operational transconductance amplifiers and fractional-order element.


2016 ◽  
pp. 71-76
Author(s):  
H. Ukhina ◽  
A. Bilenko ◽  
V. Sytnikov

The paper considers improving efficiency of NPP software based I&C during adjustment and readjustment of its characteristics. The research analyzes impact of transfer function coefficient of digital components on features of frequency-response characteristics, which shall be considered during design of software based I&C. The paper objective was to determine the numerator and denominator dependencies of transfer function of first order high-pass and low-pass digital filters of cut-off frequency, and also to determine dependencies on pulsation coefficient.


2020 ◽  
Author(s):  
Bahram Zonooz ◽  
A. John Van Opstal

AbstractChronic single-sided deaf (CSSD) listeners lack the availability of binaural difference cues to localize sounds in the horizontal plane. Hence, for directional hearing they have to rely on different types of monaural cues: loudness perceived in their hearing ear, which is affected in a systematic way by the acoustic head shadow, on spectral cues provided by the low-pass filtering characteristic of the head, and on high-frequency spectral-shape cues from the pinna of their hearing ear. Presumably, these cues are differentially weighted against prior assumptions on the properties of sound sources in the environment. The rules guiding this weighting process are not well understood. In this preliminary study, we trained three CSSD listeners to localize a fixed intensity, high-pass filtered sound source at ten locations in the horizontal plane with visual feedback. After training, we compared their localization performance to sounds with different intensities, presented in the two-dimensional frontal hemifield to their pre-training results. We show that the training had rapidly readjusted the contributions of monaural cues and internal priors, which resulted to be imposed by the multisensory information provided during the training. We compare the results with the strategies found for the acute monaural hearing condition of normal-hearing listeners, described in an earlier study [1].


1992 ◽  
Vol 82 (1) ◽  
pp. 238-258
Author(s):  
Stuart A. Sipkin ◽  
Arthur L. Lerner-Lam

Abstract The availability of broadband digitally recorded seismic data has led to an increasing number of studies using data from which the instrument transfer function has been deconvolved. In most studies, it is assumed that raw ground motion is the quantity that remains after deconvolution. After deconvolving the instrument transfer function, however, seismograms are usually high-pass filtered to remove low-frequency noise caused by very long-period signals outside the frequency band of interest or instabilities in the instrument response at low frequencies. In some cases, data must also be low-pass filtered to remove high-frequency noise from various sources. Both of these operations are usually performed using either zero-phase (acausal) or minimum-phase (causal) filters. Use of these filters can lead to either bias or increased uncertainty in the results, especially when taking integral measures of the displacement pulse. We present a deconvolution method, based on Backus-Gilbert inverse theory, that regularizes the time-domain deconvolution problem and thus mitigates any low-frequency instabilities. We apply a roughening constraint that minimizes the long-period components of the deconvolved signal along with the misfit to the data, emphasizing the higher frequencies at the expense of low frequencies. Thus, the operator acts like a high-pass filter but is controlled by a trade-off parameter that depends on the ratio of the model variance to the residual variance, rather than an ad hoc selection of a filter corner frequency. The resulting deconvolved signal retains a higher fidelity to the original ground motion than that obtained using a postprocess high-pass filter and eliminates much of the bias introduced by such a filter. A smoothing operator can also be introduced that effectively applies a low-pass filter. This smoothing is useful in the presence of blue noise, or if inferences about source complexity are to be made from the roughness of the deconvolved signal.


2012 ◽  
Vol 37 (4) ◽  
pp. 447-454
Author(s):  
James W. Beauchamp

Abstract Source/filter models have frequently been used to model sound production of the vocal apparatus and musical instruments. Beginning in 1968, in an effort to measure the transfer function (i.e., transmission response or filter characteristic) of a trombone while being played by expert musicians, sound pressure signals from the mouthpiece and the trombone bell output were recorded in an anechoic room and then subjected to harmonic spectrum analysis. Output/input ratios of the signals’ harmonic amplitudes plotted vs. harmonic frequency then became points on the trombone’s transfer function. The first such recordings were made on analog 1/4 inch stereo magnetic tape. In 2000 digital recordings of trombone mouthpiece and anechoic output signals were made that provide a more accurate measurement of the trombone filter characteristic. Results show that the filter is a high-pass type with a cutoff frequency around 1000 Hz. Whereas the characteristic below cutoff is quite stable, above cutoff it is extremely variable, depending on level. In addition, measurements made using a swept-sine-wave system in 1972 verified the high-pass behavior, but they also showed a series of resonances whose minima correspond to the harmonic frequencies which occur under performance conditions. For frequencies below cutoff the two types of measurements corresponded well, but above cutoff there was a considerable difference. The general effect is that output harmonics above cutoff are greater than would be expected from linear filter theory, and this effect becomes stronger as input pressure increases. In the 1990s and early 2000s this nonlinear effect was verified by theory and measurements which showed that nonlinear propagation takes place in the trombone, causing a wave steepening effect at high amplitudes, thus increasing the relative strengths of the upper harmonics.


2016 ◽  
Vol 15 (12) ◽  
pp. 2579-2586
Author(s):  
Adina Racasan ◽  
Calin Munteanu ◽  
Vasile Topa ◽  
Claudia Pacurar ◽  
Claudia Hebedean

2007 ◽  
Vol 16 (04) ◽  
pp. 507-516 ◽  
Author(s):  
SHAHRAM MINAEI ◽  
ERKAN YUCE

In this paper, a universal current-mode second-order active-C filter for simultaneously realizing low-pass, band-pass and high-pass responses is proposed. The presented filter employs only three plus-type second-generation current-controlled conveyors (CCCII+s). This filter needs no critical active and passive component matching conditions and no additional active and passive elements for realizing high output impedance low-pass, band-pass and high-pass characteristics. The angular resonance frequency (ω0) and quality factor (Q) of the proposed resistorless filter can be tuned electronically. To verify the theoretical analysis and to exhibit the performance of the proposed filter, it is simulated with SPICE program.


Sign in / Sign up

Export Citation Format

Share Document