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OBJECTIVE This study examined the effect of microphone placement on the interaural level differences (ILDs) available to bilateral cochlear implant (BiCI) users, and the subsequent effects on horizontal-plane sound localization. DESIGN Virtual acoustic stimuli for sound localization testing were created individually for eight BiCI users by making acoustic transfer function measurements for microphones placed in the ear (ITE), behind the ear (BTE), and on the shoulders (SHD). The ILDs across source locations were calculated for each placement to analyze their effect on sound localization performance. Sound localization was tested using a repeated-measures, within-participant design for the three microphone placements. RESULTS The ITE microphone placement provided significantly larger ILDs compared to BTE and SHD placements, which correlated with overall localization errors. However, differences in localization errors across the microphone conditions were small. CONCLUSIONS The BTE microphones worn by many BiCI users in everyday life do not capture the full range of acoustic ILDs available, and also reduce the change in cue magnitudes for sound sources across the horizontal plane. Acute testing with an ITE placement reduced sound localization errors along the horizontal plane compared to the other placements in some patients. Larger improvements may be observed if patients had more experience with the new ILD cues provided by an ITE placement.

In four experiments listeners were rotated or were stationary. Sounds came from a stationary loudspeaker or rotated from loudspeaker to loudspeaker around an azimuth array. When either sounds or listeners rotate the auditory cues used for sound source localization change, but in the everyday world listeners perceive sound rotation only when sounds rotate not when listeners rotate. In the everyday world sound source locations are referenced to positions in the environment (a world-centric reference system). The auditory cues for sound source location indicate locations relative to the head (a head-centric reference system), not locations relative to the world. This paper deals with a general hypothesis that the world-centric location of sound sources requires the auditory system to have information about auditory cues used for sound source location and cues about head position. The use of visual and vestibular information in determining rotating head position in sound rotation perception was investigated. The experiments show that sound rotation perception when sources and listeners rotate was based on acoustic, visual, and, perhaps, vestibular information. The findings are consistent with the general hypotheses and suggest that sound source localization is not based just on acoustics. It is a multisystem process.

In this letter, we propose a new auditory distance rendering (ADR) algorithm based on the interchannel phase difference (ICPD) control. In the conventional ICPD control, distance perception of the sound image is nonlinearly controlled, and directional localization of the sound image can be biased by changes of the interaural cues. These problems are caused by applying the frequency-independent ICPD without considering the acoustic transfer paths of the system setup. To solve these problems, first, the interaural cues of ear signals are analyzed by binaural auditory simulations. Then, stereophonic ADR filters are designed that produce ear signals with a linearly controlled interaural cross-correlation (IACC) and a consistent interaural level difference (IALD) for sophisticated distance perception under the given stereo setup. Subjective test results show that the proposed algorithm can provide better distance controllability than the conventional method with reduced lateralization blur of the sound image.

The aim of this letter is to address a little understood question in sound source localization: Can the distance of a near sound source affect our own perception of its elevation? The issue is studied by means of an objective analysis of a database of distance-dependent head-related transfer functions (HRTFs) of a KEMAR (Knowles Electronic Manikin for Acoustic Research) mannequin with different pinnae on a dense spatial grid. Iso-directional HRTFs are compared through spectral error metrics; results indicate significant distance-dependent HRTF modifications due to the pinna occur when the source is close to the interaural axis.

An improved sound source localization (SSL) method has been developed that is based on the generalized cross-correlation (GCC) method weighted by the phase transform (PHAT) for use with binaural robots equipped with two microphones inside artificial pinnae. The conventional SSL method based on the GCC-PHAT method has two main problems when used on a binaural robot platform: 1) diffraction of sound waves with multipath interference caused by the contours of the robot head, which affects localization accuracy, and 2) front-back ambiguity, which limits the localization range to half the horizontal space. The diffraction problem was overcome by incorporating a new time delay factor into the GCC-PHAT method under the assumption of a spherical robot head. The ambiguity problem was overcome by utilizing the amplification effect of the pinnae for localization over the entire azimuth. Experiments conducted using two dummy heads equipped with small or large pinnae showed that localization errors were reduced by 8.91° (3.21° vs. 12.12°) on average with the new time delay factor compared with the conventional GCC-PHAT method and that the success rate for front-back disambiguation using the pinnae amplification effect was 29.76 % (93.46 % vs. 72.02 %) better on average over the entire azimuth than with a conventional head related transfer function (HRTF)-based method.

Listeners localized the free-field sources of either one or two simultaneous and independently generated noise bursts. Listeners' localization performance was better when localizing one rather than two sound sources. With two sound sources, localization performance was better when the listener was provided prior information about the location of one of them. Listeners also localized two simultaneous noise bursts that had sinusoidal amplitude modulation (AM) applied, in which the modulation envelope was in-phase across the two source locations or was 180° out-of-phase. The AM was employed to investigate a hypothesis as to what process listeners might use to localize multiple sound sources. The results supported the hypothesis that localization of two sound sources might be based on temporal-spectral regions of the combined waveform in which the sound from one source was more intense than that from the other source. The interaural information extracted from such temporal-spectral regions might provide reliable estimates of the sound source location that produced the more intense sound in that temporal-spectral region.

Multisensory interactions involve coordination and sometimes competition between multiple senses. Vision usually dominates audition in spatial judgments when light and sound stimuli are presented from two different physical locations. This study investigated the influence of vision on the perceived location of a phantom sound source placed in a stereo sound field using a pair of loudspeakers emitting identical signals that were delayed or attenuated relative to each other. Results show that although a similar horizontal range (+/−45°) was reported for timing-modulated and level-modulated signals, listeners' localization performance showed greater variability for the timing signals. When visual stimuli were presented simultaneously with the auditory stimuli, listeners showed stronger visual bias for timing-modulated signals than level-modulated and single-speaker control signals. Trial-to-trial errors remained relatively stable over time, suggesting that sound localization uncertainty has an immediate and long-lasting effect on the across-modal bias. Binaural signal analyses further reveal that interaural differences of time and intensity—the two primary cues for sound localization in the azimuthal plane—are inherently more ambiguous for signals placed using timing. These results suggest that binaural ambiguity is intrinsically linked with localization variability and the strength of cross-modal bias in sound localization.

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Binaural sound source localization can be performed by imitation of the fundamental mechanisms of the human auditory system, which is based on the integrated effects of ear, pinnae, head and torso. In particular, two physical cues can be exploited, i.e. the Interaural Time Difference (ITD) and the Interaural Level Difference (ILD). It is known that joint use of ITD and ILD provides good source azimuth estimations. In many practical situations binaural localization has to be performed in closed environments, where the presence of reverberation degrades the performance of available position estimators. In this paper a possible solution to this difficult problem is introduced. The proposed solution is based on proper use of cepstral prefiltering prior to source localization by ITD and ILD. It is shown that cepstrum can help in reducing the effects of reverberation, thus yielding better location estimates.

Users of a cochlear implant together with a contralateral hearing aid - so-called bimodal listeners - have difficulties with localizing sound sources. This is mainly due to the distorton of interaural time and level difference cues (ITD and ILD), and limited ITD sensitivity. An algorithm is presented that enhances ILD cues. Horizontal plane sound-source localization performance of 6 bimodal listeners was evaluated in (1) a real sound field with their clinical devices, (2) in a virtual sound field, under direct computer control and (3) in a virtual sound field with ILD enhancement. The results in the real sound field did not differ significantly from the results in the virtual field, and ILD enhancement improved localization performance by 4-10 degrees absolute error, relative to a mean absolute error of 28 degrees in the condition without ILD enhancement. ispartof: Journal of the Acoustical Society of America vol:130 issue:5 pages:2817-2826 ispartof: location:United States status: published

This article discusses the effect of frequency bandwidth on the maximum absolute value of interaural cross-correlation coefficient (IACC) in relation to sound image width (also known as auditory source width, ASW). Subjective experiments concerning sound image width were performed using reproduced sounds of a violin. This article shows that separation into 1/24 and more octave bands and averaging IACC values of these frequency bands leads to a higher correlation to sound image width than IACC in broader frequency bands. Moreover the average of 1/ n octave bands of the temporal deviation of interaural time differences and interaural level differences could be confirmed as measures correlating with sound image width.

In this paper, equalization zone of two crosstalk cancellation (CC) schemes, which are the one based on only head related transfer function (HRTF) and the other one based on interaural intensity/time difference (ITD/IID) as well as HRTF is studied. To do this, the condition numbers and ITD/IID levels of two schemes are shown.

We propose a novel design of an artificial robot ear for sound direction estimation using two measured outputs only. The spectral features in the interaural transfer functions (ITFs) of the proposed artificial ears are distinctive and move monotonically according to the sound direction. Thus, these features provide effective sound cues to estimate sound direction using the measured two output signals. Bilateral asymmetry of microphone positions can enhance the estimation performance even in the median plane where interaural differences vanish. We propose a localization method to estimate the lateral and vertical angles simultaneously. The lateral angle is estimated using interaural time difference and Woodworth and Schlosberg's formula, and the front-back discrimination is achieved by finding the spectral features in the ITF estimated from two measured outputs. The vertical angle of a sound source in the frontal region is estimated by comparing the spectral features in the estimated ITF with those in the database built in an anechoic chamber. The feasibility of the designed artificial ear and the estimation method were verified in a real environment. In the experiment, it was shown that both the front-back discrimination and the sound direction estimation in the frontal region can be achieved with reasonable accuracy. Thus, we expect that robots with the proposed artificial ear can estimate the direction of speaker from two output signals only.

In this paper an analysis of moving sound source localization via headphones is presented. Also the influence of the inter-click interval on this localization is studied. The experimental sound is a short delta sound of 5 ms, generated for the horizontal frontal plane, for distances from 0.5 m to 5 m and azimuth of 32° to both left and right sides with respect of the middle line of the listener head convolutioned with individual HRTFs. The results indicate that the best accurate localization was achieved for the ICI of 150 ms. Comparing the localization accuracy in distance and azimuth is deduced that the best results have been achieved for azimuth. The results show that the listeners are able to extract accurately the distance and direction of the moving sound for larger inter-click intervals.

Using our developed acoustical telepresence robot, TeleHead, we have so far confirmed that not only stationary binaural features, but also dynamic cues from head movement play important roles in sound localization. In this study, aiming towards the realization of an ideal acoustical telepresence robot, we clarify the relation between the head movement and the accuracy of sound localization in sound localization experiments. We examined two factors related to head movement that should have an impact on sound localization accuracy: observation from multiple postures and dynamic information during head movement. The results suggest that both factors improve the accuracy of sound localization in experiments. Moreover, even when we can use only one of these factors, the accuracy of sound localization is almost the same as the subject's original accuracy. The results confirm that even under very bad communication, control and head-shape conditions, the synchronization of head movement is important for building ...

This paper presents a mathematical model for sound localization in space using two-microphone devices that possess at least two degrees of freedom. It proves a series of theorems and lemmas that are based on time difference of arrival measurements and movements of the interaural axis, forming a powerful instrument for practical robot applications. For instance, it shows that a single determined rotation of the interaural axis is sufficient to exactly yield the azimuth or the elevation of an immobile sound source in the far field, independently of microphone spacing and the speed of sound and hence of the surrounding medium. It proves that at any moment the knowledge of one value determines the magnitude of the other, with the restriction that the sign of the second value is undefined, which means that, depending on the rotation, either the back-front or the up-down ambiguity is kept unsolved. This paper also shows that parallax motion unlocks essential information about the distance and the Cartesian coordinates of the sound source. Shifting the microphone system sideways fixes the distance and the coordinate on the interaural axis. Combining rotation and translation movements completely solves the localization problem. In order to illustrate the efficacy of the model, this paper presents experiments with a low cost robot developer kit during which the azimuth, the elevation, and the distance of continuous sound sources are determined at a precision of 10 degrees and 0.5 m, respectively. Achieving this performance with low power material demonstrates how easily the model can be implemented into any robotic system.

There are two general neurophysiological models of sound lateralization mechanisms which may be active in man. Both of the models are derived from studies in animals (one in barn owls, and one in mammals), and both have displayed some weakness in generalizability. One model advocates a population of neurons narrowly tuned to different interaural disparity values across the behaviorally relevant range, so that the cue value, and therefore the source azimuth, is represented by which neurons of the array are activated by the stimulus. The second model posits the existence of only two neural channels, each broadly tuned to interaural cue values favoring one acoustic hemifield, so that, especially for sources near the midline, cue value and therefore source azimuth is encoded by the relative activation of the two neural populations. The present article reviews three recent psychophysical studies, each using selective adaptation paradigms to probe sound lateralization mechanisms based on interaural disparities in normal human listeners. These experiments provided evidence on the frequency-specificity of interaural disparity coding and revealed its sensitivity to recent stimulus history. The data from those studies, however, also help distinguish the two lateralization models, and favor a perceptual architecture for sound lateralization in man based on the activity of two, hemifield-tuned azimuthal channels.

This paper presents a sound source localization algorithm based on zero crossings in acoustic reverberant environments. From sudden increases of acoustic energy in a non-stationary signal, we identify intervals which dominantly include signal components through direct paths from a source to sensors because they may provide reliable localization cues corresponding to the source direction despite the reverberation. Experimental results show that the presented algorithm which is based on zero crossings with the detection of direct components efficiently accomplishes sound source localization in reverberant environments.

This paper presents a novel real-time robotic binaural sound localization method based on hierarchical fuzzy artificial neural networks and a generic set of head related transfer functions. The robot is a humanoid equipped with the KEMAR artificial head and torso. Inside the ear canals two small microphones play the role of the eardrums in collecting the impinging sound waves. The neural networks are trained using synthesized sound sources placed every 5° from 0° to 255° in azimuth, and every 5° from − 45° to 80° in elevation. To improve generalization, the training data was corrupted with noise. Thanks to fuzzy logic, the method is able to interpolate at its output, locating with high accuracy sound sources at positions which were not used for training, even in presence of strong distortion. In order to achieve high localization accuracy, two different binaural cues are combined, namely the interaural intensity differences and interaural time differences. As opposed to microphone-array methods, the presented technique, uses only two microphones to localize sound sources in a real-time 3D environment.

The head-related transfer functions (HRTFs) describe individual acoustic transformation that sound sources undergo due to human anatomy before arriving at the left and right tympanic membranes. The resulting spectral modifications are the main localization cues for elevation detection in space. In this paper, synthetic HRTF mod- els able to render the vertical spatial dimension in virtual auditory displays, are evaluated via auditory models. Perceptually-motivated metrics describe the output of 4 virtual experiments that numeri- cally simulate real listening experiments for 20 virtual subjects. The current implementation considers a limited set of parameters for a structural model of the pinna acting as a proof-of-concept of such approach. Accordingly, results confirm that the research framework is a flexible tool for systematic evaluation of different instances of structural model.

This paper proposes a biologically inspired and technically implemented sound localization system to robustly estimate the position of a sound source in the frontal azimuthal half-plane. For localization, binaural cues are extracted using cochleagrams generated by a cochlear model that serve as input to the system. The basic idea of the model is to separately measure interaural time differences and interaural level differences for a number of frequencies and process these measurements as a whole. This leads to two-dimensional frequency versus time-delay representations of binaural cues, so-called activity maps. A probabilistic evaluation is presented to estimate the position of a sound source over time based on these activity maps. Learned reference maps for different azimuthal positions are integrated into the computation to gain time-dependent discrete conditional probabilities. At every timestep these probabilities are combined over frequencies and binaural cues to estimate the sound source position. In addition, they are propagated over time to improve position estimation. This leads to a system that is able to localize audible signals, for example human speech signals, even in reverberating environments.

The robot has evolved to meet various needs in the industrial world and the home flexibly today. However, an intellectual robot has not been developed yet. In the future, the intellectual robot that will think for myself, judges, and acts is needed. One of the important senses added to the robot is aural. It is said, “Sound Localization” to presume the direction of coming among these. It aims at the achievement of three dimension sound localization with two microphones that assume the application of the humanoid robot the auditory system in this research. Then, binaural model using dummy head was made as man's auditory model. In this research, the sound source direction is decided by two angles, one is the horizontal angle α, and the other is vertical angle β. The former is presumed from the difference of the time of arrival of the sound from the source to a right and left ear, and the latter is presumed from spectrum correlation analysis using HRTF (Head Related Transfer Function). HRTF is a transfer function of spectrum modulation caused because the sound is diffracted and reflects by the ruggedness of the head and the earlobe. The directional information of the sound is taken out by using information that system has as knowledge. The direction location experiment was done in the anechoic chamber for some sources. The correct presumption rate became about 90% on the average, and was able to show the possibility of the sound localization by this technique.

Binaural processing in normal hearing activities is based on the ability of listeners to use the information provided by the differences between the signals at the two ears. The most prominent differences are the interaural time difference and the interaural level difference, both of which depend on frequency. This paper describes the stages by which these differences are estimated by the physiological structures of the auditory system, summarizes the sensitivity of the human listener to these differences, and reviews the nature of the interaural differences in realistic environments.

Auditory motion can be simulated by presenting binaural sounds with time-varying interaural time delays. Human cortical responses to the rate of auditory motion were studied by recording auditory evoked magnetic fields with a 122-channel whole-head magnetometer. Auditory motion from central to right and then to central was produced by varying interaural time differences between ears. The results showed that the N1m latencies and amplitudes were not affected by the fluctuation of interaural time delay; however, the peak amplitude of P2m significantly increased as a function of fluctuation of the interaural time delay.

The purpose of this study was to investigate sound localization in subjects bilaterally implanted with MED-EL COMBI 40/40+ cochlear implants. In addition, the sensitivity to interaural cues was assessed.In the localization test (11 subjects), CCITT noise (500 ms, original and HRTF-filtered, 70/75/80 dB sound pressure level) was presented from one of seven loudspeakers between -90 degrees and 90 degrees azimuth. The subject had to indicate which loudspeaker the noise was presented from. Sensitivity to interaural level differences (ILD) was assessed by performing localization tests (4 subjects) with the loudness of the two speech processors unbalanced to various degrees. To investigate the subjects' sensitivity to interaural time differences (ITD), lateralization was measured (7 subjects) as a function of the time difference between two Gaussian-like pulses, each directed to one of the subject's speech processor microphones by way of headphones.The judgments of all subjects significantly correlated with the positions of the loudspeakers. The scatter in the judged azimuth measured by the standard deviation of the responses was on average 27.5 degrees . Unbalanced loudness of the speech processors produced a bias in azimuth toward the speech processor with the louder volume setting. The mean rate of shift was 1.4 degrees per unit on the Wuerzburg loudness scale. Six of seven subjects showed a significant sensitivity to ITDs with the approximate time difference required for complete lateralization being 1,200 micros on an average. The one subject not showing a sensitivity to ITDs performed worst in the localization test.Bilateral cochlear implantation can restore spatial hearing in cochlear implant users. Both ILDs and ITDs are used by bilateral cochlear implant users in sound localization with ILDs appearing to be the dominant cue.

This study investigates whether the salience of the pitch associated with a single reflection of a broadband sound, such as noise, is determined by the monaural information mediated by the stimuli at the two ears, or by the relative locations of the primary sound and the reflection. Pitch strength was measured as a function of the reflection delay and the lateral displacement between the primary sound and the reflection. Thereby, lateral displacement was produced by means of interaural time differences (ITDs) in experiment 1 and interaural level differences (ILDs) in experiment 3. The results from both experiments are in accordance with the assumption that the strength of the pitch associated with a reflection is based on a central average of the internal representations of the stimuli at the two ears. This notion was corroborated by experiment 2, which showed that the results from experiment 1 could be mimicked by simply adding the stimuli from the two ears and presenting the merged stimulus identically to both ears.

Horizontal localization is based on extraction of the interaural time and level differences (ITD, ILD). Even in complex scenes with multiple talkers and reverberation, the auditory system is remarkably good at estimating the individual talker locations. Several previous models proposed mechanisms that stressed the importance of ITD in the localization process. Here, we examine whether azimuth estimation in complex scenarios can be based solely on ILD. We implemented a model (based on Faller and Merimaa, 2004) in which azimuth estimation was based on ILDs of signal parts with high interaural correlation and with spectral profile matching that of the target. Comparison with experimental data (Kopco et al., 2010) showed that highly correlated parts of the signal, if available, provide reliable ILD estimates sufficient for precise target localization. However, for lateral target positions, at which the target dominates one ear but not the other, interaural correlation was too low to guide ILD extraction. In such cases, a new model based on finding maximum ILD provided good estimates even if maskers dominated in the worse ear. The combined model predictions matched the experimental data with target locations between -50° and 50° and for 4 maskers in reverberation. [Work supported by APVV-0452-12, H2020-MSCA-RISE-2015 #691229.]Horizontal localization is based on extraction of the interaural time and level differences (ITD, ILD). Even in complex scenes with multiple talkers and reverberation, the auditory system is remarkably good at estimating the individual talker locations. Several previous models proposed mechanisms that stressed the importance of ITD in the localization process. Here, we examine whether azimuth estimation in complex scenarios can be based solely on ILD. We implemented a model (based on Faller and Merimaa, 2004) in which azimuth estimation was based on ILDs of signal parts with high interaural correlation and with spectral profile matching that of the target. Comparison with experimental data (Kopco et al., 2010) showed that highly correlated parts of the signal, if available, provide reliable ILD estimates sufficient for precise target localization. However, for lateral target positions, at which the target dominates one ear but not the other, interaural correlation was too low to guide ILD extraction. In s...

The precedence effect is the observation that, when a sound is repeated after a delay of a few milliseconds, the interaural cues of the leading sound have greater influence on localization and lateralization judgments of the composite sound than do the cues of the lagging sound. Classic studies of the precedence effect with transient stimuli have suggested auditory mechanisms that emphasize earlier-arriving cues. More recent studies have shown, however, that non-transient sounds—steady, long-duration noises with slow onset—also give rise to the precedence effect. This ongoing precedence effect is difficult to explain with the abrupt-onset mechanisms invoked to explain the effect with transients. The present model of the ongoing effect assumes that listeners use interaural cues within auditory bands to form band-specific lateral positions, which are then integrated across bands. Within-band positions are estimated from the interaural level difference and the interaural phase during the rising side of bandpass envelope modulations. Cross-band integration includes a mechanism that favors cues that are consistent across wider frequency regions. Predictions from the model will be compared to published data. [Work supported by NIDCD R01 01625.]The precedence effect is the observation that, when a sound is repeated after a delay of a few milliseconds, the interaural cues of the leading sound have greater influence on localization and lateralization judgments of the composite sound than do the cues of the lagging sound. Classic studies of the precedence effect with transient stimuli have suggested auditory mechanisms that emphasize earlier-arriving cues. More recent studies have shown, however, that non-transient sounds—steady, long-duration noises with slow onset—also give rise to the precedence effect. This ongoing precedence effect is difficult to explain with the abrupt-onset mechanisms invoked to explain the effect with transients. The present model of the ongoing effect assumes that listeners use interaural cues within auditory bands to form band-specific lateral positions, which are then integrated across bands. Within-band positions are estimated from the interaural level difference and the interaural phase during the rising side of bandp...

The conceptualization and implementation of a psychoacoustic sound field synthesis system for music is presented. Critical bands, the precedence effect, and integration times of the auditory system as well as the radiation characteristics of musical instruments are implemented in the signal processing. Interaural coherence, masking and auditory scene analysis principles are considered as well. The sound field synthesis system creates a natural, spatial sound impression and precise source localization for listeners in extended listening area, even with a low number of loudspeakers. Simulations and a listening test provide a proof of concept. The method is particularly robust for signals with impulsive attacks and quasi-stationary phases as in the case of many instrumental sounds. It is compatible with many loudspeaker setups, such as 5.1, ambisonics systems and loudspeaker arrays for wave front synthesis. The psychoacoustic sound field synthesis approach is an alternative to physically centered wave field ...

Binaural source localization is a popular technique in various applications, such as hearing aids, mobile robot, video conference, etc. However, robust binaural cues estimate and suitable localization strategy are always limiting its performance. In this paper, a new algorithm for binaural sound source localization is presented, which is a two-layer model. In the first layer, a spectral weighting generalized cross correlation-phase transformation (GCC-PHAT) method is presented for robust time-delay estimation, by which the probabilistic azimuths of sound source are obtained. In the second layer, an improved algorithm is introduced, which is named Compensated Interaural Intensity Difference (CIID). Based on the probabilistic azimuth localization results and CIID features, spatial grid matching (SGM) is presented to provide a Bayesian model for localizing azimuth and elevation. Compared with three other algorithms, experimental results show that the proposed method has a robust result even in noisy environments.

The role of interaural differences in localization of spatially segregated sound was clarified by performing listening tests on sound localization. Four subjects were asked to distinguish two sources of white noise with various interaural time differences (ITDs) or interaural level differences (ILDs). As a result, the segregated rate reached 80% when the ITD between two white noise sources corresponded to an angular difference of 22 degrees. The segregated rate degraded to 50% when the ITD between two white noise sources corresponded to an angular difference of 7 degrees. In addition, the segregated rate reached 80% when the ILD between two white noise sources corresponded to an angular difference of 30 degrees. The segregated rate degraded to 50% when the ILD between two white noise sources corresponded to an angular difference of about 15 degrees. These results suggest that sound sources could be spatially segregated using even only ITD or ILD.

In this report, we used filtered noise bands to constrain listeners' access to interaural level differences (ILDs) and interaural time differences (ITDs) in a sound source localization task. The samples of interest were listeners with single-sided deafness (SSD) who had been fit with a cochlear implant in the deafened ear (SSD-CI). The comparison samples included listeners with normal hearing and bimodal hearing, i.e. with a cochlear implant in 1 ear and low-frequency acoustic hearing in the other ear. The results indicated that (i) sound source localization was better in the SSD-CI condition than in the SSD condition, (ii) SSD-CI patients rely on ILD cues for sound source localization, (iii) SSD-CI patients show functional localization abilities within 1-3 months after device activation and (iv) SSD-CI patients show better sound source localization than bimodal CI patients but, on average, poorer localization than normal-hearing listeners. One SSD-CI patient showed a level of localization within normal limits. We provide an account for the relative localization abilities of the groups by reference to the differences in access to ILD cues.

Reflections can play an important role in human perception of sound. While they can positively contribute to the perceived sound quality, they may also interfere with the reproduction of, for example, crosstalk cancelled binaural sounds through loudspeakers. In this paper, we study the influence of 1st and 2nd order reflections on a crosstalk-cancelled desktop reproduction system, through an analysis of the interaural time differences. The direct and reflected sounds are calculated using an image-source model. Meanwhile, the crosstalk cancellation filters are calculated assuming anechoic conditions, and therefore proper sound reproduction is now questionable. In this scenario, the reflections are found to introduce changes in the interaural phase differences and in the interaural group delay. These changes are analyzed and the possible effect on sound localization is investigated using a subjective localization experiment. The results indicated that for the studied setup, the localization accuracy was, practically, unaffected by the low order reflections.

A computational model of auditory localization resulting in performance similar to humans is reported. The model incorporates both the monaural and binaural cues available to a human for sound localization. Essential elements used in the simulation of the processes of auditory cue generation and encoding by the nervous system include measured head-related transfer functions (HRTFs), minimum audible field (MAF), and the Patterson-Holdsworth cochlear model. A two-layer feed-forward back-propagation artificial neural network (ANN) was trained to transform the localization cues to a two-dimensional map that gives the direction of the sound source. The model results were compared with (i) the localization performance of the human listener who provided the HRTFs for the model and (ii) the localization performance of a group of 19 other human listeners. The localization accuracy and front-back confusion error rates exhibited by the model were similar to both the single listener and the group results. This suggests that the simulation of the cue generation and extraction processes as well as the model parameters were reasonable approximations to the overall biological processes. The amplitude resolution of the monaural spectral cues was varied and the influence on the model's performance was determined. The model with 128 cochlear channels required an amplitude resolution of approximately 20 discrete levels for encoding the spectral cue to deliver similar localization performance to the group of human listeners.

When sound sources or listeners rotate, the acoustic cues used for sound source localization change, but in the everyday world listeners perceive sound rotation only when the sound source rotates not when the listener rotates. That is, in the everyday world, sound source locations are referenced to positions in the environment (a world-centric reference system). The acoustic cues for sound source location indicate a sound source’s location relative to the head (a head-centric reference system), not locations relative to the world. To compute world-centric locations of sound sources, the auditory spatial system must have information about the acoustic cues used for sound source location and cues about the position of the head. The use of visual and vestibular information in determining head position in sound rotation perception was investigated in four experiments. The experiments clearly show, for the first time, that sound source localization when sound sources and listeners rotate is based on acoustic and visual, and sometimes vestibular information. Sound source localization is not based just on acoustics. It is a multisensory process. [Research supported by an AFOSR grant.]

The detection of interaural time differences (ITDs) for sound localization critically depends on the similarity between the left and right ear signals (interaural correlation). We show that, like humans, owls can localize phantom sound sources well until the correlation declines to a very low value, below which their performance rapidly deteriorates. Decreasing interaural correlation also causes the response of the owl’s tectal auditory neurons to decline nonlinearly, with a rapid drop followed by a more gradual reduction. A detection-theoretic analysis of the statistical properties of neuronal responses could account for the variance of behavioral responses as interaural correlation is decreased. Finally, cross-correlation analysis suggests that low interaural correlations cause misalignment of cross-correlation peaks across different frequencies, contributing heavily to the nonlinear decline in neural and ultimately behavioral performance.

Illusory sensations of posture and motion can be perceived in some situations. For example, visually induced self-motion has been studied, while we can demonstrate self-motion and body sway caused by virtually moving sound-sources. We investigated such self-motion and body sway. The center of pressure and the sensation level of self-motion were measured and analyzed to discuss their characteristics in terms of the interaural time difference (ITD) and interaural level difference (ILD) caused by moving sound. As a result, periodic body sway synchronized with sound source movement was observed. Larger ITD and ILD had an influence on self-motion and body sway. In addition, azimuths for ITD and ILD did not need to be matched. Characteristics of auditory induced body sway might be expected to apply for healthcare and fitness fields.