Knowles Electronics Manikin For Acoustic Research Research Articles

The possible range of interaural cross correlation after cochlear implant processing

Normal hearing listeners rely on their binaural auditory system to combine signals from both ears into single auditory objects, which aids source segregation and sound localization. The ability to combine signals from the two ears is influenced by interaural symmetry - the similarity of the place of stimulation along the left and right cochlea—and the interaural cross-correlation (ICC) of the signal, calculated as the peak correlation between the signal for the two ears across unilateral temporal offsets. The ability of bilateral cochlear implant (CI) users to combine signals from the two ears can be worse than that of NH listeners. This is partly attributed to decreases in interaural symmetry related to factors such as differences in electrode array insertion depth. A secondary effect of interaural asymmetry is that it can result in reduced ICC at interaurally symmetric locations, mediated by aspects of CI processing. This study aims to investigate how CI processing and interaural asymmetry affect the ICC available to CI users for interaurally symmetric locations, using electrodograms recorded from a CI processor placed on a Knowles Electronics Manikin for Acoustic Research (KEMAR). Inferences on how ICC is altered with interaural asymmetry for CI users will be discussed.

Effects of Directional Microphone and Noise Reduction on Subcortical and Cortical Auditory-Evoked Potentials in Older Listeners With Hearing Loss.

Understanding how signal processing influences neural activity in the brain with hearing loss is relevant to the design and evaluation of features intended to alleviate speech-in-noise deficits faced by many hearing aid wearers. Here, we examine whether hearing aid processing schemes that are designed to improve speech-in-noise intelligibility (i.e., directional microphone and noise reduction) also improve electrophysiological indices of speech processing in older listeners with hearing loss. The study followed a double-blind within-subjects design. A sample of 19 older adults (8 females; mean age = 73.6 years, range = 56-86 years; 17 experienced hearing aid users) with a moderate to severe sensorineural hearing impairment participated in the experiment. Auditory-evoked potentials associated with processing in cortex (P1-N1-P2) and subcortex (frequency-following response) were measured over the course of two 2-hour visits. Listeners were presented with sequences of the consonant-vowel syllable /da/ in continuous speech-shaped noise at signal to noise ratios (SNRs) of 0, +5, and +10 dB. Speech and noise stimuli were pre-recorded using a Knowles Electronics Manikin for Acoustic Research (KEMAR) head and torso simulator outfitted with hearing aids programmed for each listener's loss. The study aid programs were set according to 4 conditions: (1) omnidirectional microphone, (2) omnidirectional microphone with noise reduction, (3) directional microphone, and (4) directional microphone with noise reduction. For each hearing aid condition, speech was presented from a loudspeaker located at 1 m directly in front of KEMAR (i.e., 0° in the azimuth) at 75 dB SPL and noise was presented from a matching loudspeaker located at 1 m directly behind KEMAR (i.e., 180° in the azimuth). Recorded stimulus sequences were normalized for speech level across conditions and presented to listeners over electromagnetically shielded ER-2 ear-insert transducers. Presentation levels were calibrated to match the output of listeners' study aids. Cortical components from listeners with hearing loss were enhanced with improving SNR and with use of a directional microphone and noise reduction. On the other hand, subcortical components did not show sensitivity to SNR or microphone mode but did show enhanced encoding of temporal fine structure of speech for conditions where noise reduction was enabled. These results suggest that auditory-evoked potentials may be useful in evaluating the benefit of different noise-mitigating hearing aid features.

A Hybrid Algorithm for Predicting Median-Plane Head-Related Transfer Functions from Anthropometric Measurements

Since head-related transfer functions (HRTFs) represent the interactions between sounds and physiological structures of listeners, anthropometric parameters represent a straightforward way to customize (or predict) individualized HRTFs. This paper proposes a hybrid algorithm for predicting median-plane individualized HRTFs using anthropometric parameters. The proposed hybrid algorithm consists of three parts: decomposition of HRTFs; selection of key anthropometric parameters; and establishing a prediction formula. Firstly, an independent component analysis (ICA) is applied to median-plane HRTFs from multiple subjects to obtain independent components and subject-dependent weight coefficients. Then, a factor analysis is used to select key anthropometric parameters relevant to HRTFs. Finally, a regression formula that connects ICA weight coefficients to key anthropometric parameters is established by a multiple linear regression. Further, the effectiveness of the proposed hybrid algorithm is verified by an objective evaluation via spectral distortion and a subjective localization experiment. The results show that, when compared with generic Knowles Electronics Manikin for Acoustic Research (KEMAR) HRTFs, the spectral characteristics of the predicted HRTFs are closer to those of the individualized HRTFs. Moreover, the predicted HRTFs can alleviate front–back and up–down confusion and improve the accuracy of localization for most subjects.

Influence of voxelization on finite difference time domain simulations of head-related transfer functions.

The scattering around the human pinna that is captured by the Head-Related Transfer Functions (HRTFs) is a complex problem that creates uncertainties in both acoustical measurements and simulations. Within the simulation framework of Finite Difference Time Domain (FDTD) with axis-aligned staircase boundaries resulting from a voxelization process, the voxelization-based uncertainty propagating in the HRTF-captured sound field is quantified for one solid and two surface voxelization algorithms. Simulated results utilizing a laser-scanned mesh of Knowles Electronics Manikin for Acoustic Research (KEMAR) show that in the context of complex geometries with local topology comparable to grid spacing such as the human pinna, the voxelization-related uncertainties in simulations emerge at lower frequencies than the generally used accuracy bandwidths. Numerical simulations show that the voxelization process induces both random error and algorithm-dependent bias in the simulated HRTF spectral features. Frequencies fr below which the random error is bounded by various dB thresholds are estimated and predicted. Particular shortcomings of the used voxelization algorithms are identified and the influence of the surface impedance on the induced errors is studied. Simulations are also validated against measurements.

High-Frequency Audibility: The Effects of Audiometric Configuration, Stimulus Type, and Device

For the last decade, the importance of providing amplification up to 9-10 kHz has been supported by multiple studies involving children and adults. The extent to which a listener with hearing loss can benefit from bandwidth expansion is dependent on the audibility of high-frequency cues. The American National Standards Institute (ANSI) devised a standard method for measuring and reporting hearing aid bandwidth for quality-control purposes. However, ANSI bandwidth measurements were never intended to reflect the true frequency range that is audible for a speech stimulus for a person with hearing loss. The purpose of this study was to (1) determine the maximum audible frequency of conventional hearing aids using a speech signal as the input through the hearing aid microphone for different degrees of hearing loss, (2) examine how the maximum audible frequency changes when the input stimulus is presented through hearing assistance technology (HAT) systems with cross-coupling of manufacturers' transmitters and receivers, and (3) evaluate how the maximum audible frequency compares with the upper limit of the ANSI bandwidth measure. Eight behind-the-ear hearing aids from five hearing aid manufacturers were selected based on a range of ANSI bandwidth upper frequency limits. Three audiometric configurations with varied degrees of high-frequency hearing loss were programmed into each hearing aid. Hearing aid responses were measured with the International Speech Test Signal (ISTS), broadband noise, and a short speech token (/asa/) as stimuli presented through a loudspeaker. HAT devices from three manufacturers were used to create five HAT scenarios. These instruments were coupled to the hearing aid programmed for the audiogram that provided the highest maximum audible frequency in the hearing aid analysis. The response from each HAT scenario was obtained using the same three stimuli as during the hearing aid analysis. All measurements were collected in an audiometric sound booth on a Knowles Electronic Manikin for Acoustic Research (KEMAR). A custom computer program was used to record responses from KEMAR. Maximum audible frequency was defined as the highest point where the Long-Term Average Speech Spectrum (LTASS) intersected the audiogram. The average maximum audible frequency measured through KEMAR ranged from 3.5 kHz to beyond 8 kHz and varied significantly across devices, audiograms, and stimuli. The specified upper limit of the ANSI bandwidth was not predictive of the maximum audible frequency across conditions. For most HAT systems, the maximum audible frequency for the hearing aid plus HAT condition was equivalent to the hearing aid for the same measurement configuration. In some cases, however, the HAT system imposed a lower maximum audible frequency than the hearing aid-only condition. The maximum audible frequency of behind-the-ear hearing aids is dependent on the degree of hearing loss, amplification device, and stimulus input. Estimating the maximum audible frequency by estimating the frequency where the speech spectrum intersects the audiogram in the high frequencies can assist clinicians in making decisions about which device or configuration of devices provides the greatest access to high-frequency information, as well as whether frequency-lowering technology should be used.

On distance dependence of pinna spectral patterns in head-related transfer functions.

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.

Aging Effects on Detection of Spectral Changes Induced by a Break in Sound Correlation

Previous studies have shown that both younger adults and older adults with clinically normal hearing are able to detect a break in correlation (BIC) between interaurally correlated sounds presented over headphones. This ability to detect a BIC improved when the correlated sounds were presented over left and right loudspeakers rather than over left and right headphones, suggesting that additional spectral cues provided by comb filtering (caused by interference between the two channels) facilitate detection of the BIC. However, older adults receive significantly less benefit than younger adults from a switch to loudspeaker presentation. It is hypothesized that this is a result of an age-related reduction in the sensitivity to the monaural spectral cues provided by comb filtering. Two experiments were conducted in this study. Correlated white noises with a BIC in the temporal middle were presented from two spatially separated loudspeakers (positioned at ±45-degree azimuth) and recorded at the right ear of a Knowles Electronic Manikin for Acoustic Research (KEMAR). In Experiment 1, the waveforms recorded at the KEMAR's right ear were presented to the participant's right ear over a headphone in 14 younger adults and 24 older adults with clinically normal hearing. In Experiment 2, 8 of the 14 younger participants participated. Under the monaurally cueing condition, the waveforms recorded at the KEMAR's right ear were presented to the participant's right ear as Experiment 1. Under the binaurally cueing condition, waveforms delivered from the left loudspeaker and those from the right loudspeaker were recorded at the KEMAR's left and right ear, respectively, thereby eliminating the spectral ripple cue, and were presented to the participant's left and right ears, respectively. For each of the two experiments, the break duration threshold for detecting the BIC was examined when the interloudspeaker interval (delay) (ILI) was 0, 1, 2, or 4 msec (left loudspeaker leading). In Experiment 1, both younger participants and older participants detected the BIC in the waveforms recorded by the right ear of KEMAR, but older participants had higher detection thresholds than younger participants when the ILI was 0, 2, or 4 msec without an effect of SPL shift between 59 and 71 dB. In Experiment 2, each of the eight younger participants was able to detect the occurrence of the BIC in either the monaurally cueing or binaural-cueing condition. In addition, the detection threshold under the monaurally cueing condition was substantially the same as that under the binaurally cueing condition at each of the four ILIs. Younger adults and older adults with clinically normal hearing are able to detect the monaural spectral changes arising from comb filtering when a sudden drop in intersound correlation is introduced. However, younger adults are more sensitive than older adults are, at detecting the BIC. The findings suggest that older adults are less able than younger adults to detect a periodic ripple in the sound spectrum. This age-related ability reduction may contribute to older adults' difficulties in hearing under noisy, reverberant conditions.

Are Two Ears Not Better Than One?

The decision to fit one or two hearing aids in individuals with binaural hearing loss has been debated for years. Although some 78% of U.S. hearing aid fittings are binaural (Kochkin , 2010), Walden and Walden (2005) presented data showing that 82% (23 of 28 patients) of their sample obtained significantly better speech recognition in noise scores when wearing one hearing aid as opposed to two. To conduct two new experiments to fuel the monaural/binaural debate. The first experiment was a replication of Walden and Walden (2005), whereas the second experiment examined the use of binaural cues to improve speech recognition in noise. A repeated measures experimental design. Twenty veterans (aged 59-85 yr), with mild to moderately severe binaurally symmetrical hearing loss who wore binaural hearing aids were recruited from the Audiology Department at the Bay Pines VA Healthcare System. Experiment 1 followed the procedures of the Walden and Walden study, where signal-to-noise ratio (SNR) loss was measured using the Quick Speech-in-Noise (QuickSIN) test on participants who were aided with their current hearing aids. Signal and noise were presented in the sound booth at 0° azimuth under five test conditions: (1) right ear aided, (2) left ear aided, (3) both ears aided, (4) right ear aided, left ear plugged, and (5) unaided. The opposite ear in (1) and (2) was left open. In Experiment 2, binaural Knowles Electronics Manikin for Acoustic Research (KEMAR) manikin recordings made in Lou Malnati's pizza restaurant during a busy period provided a typical real-world noise, while prerecorded target sentences were presented through a small loudspeaker located in front of the KEMAR manikin. Subjects listened to the resulting binaural recordings through insert earphones under the following four conditions: (1) binaural, (2) diotic, (3) monaural left, and (4) monaural right. Results of repeated measures ANOVAs demonstrated that the best speech recognition in noise performance was obtained by most participants with both ears aided in Experiment 1 and in the binaural condition in Experiment 2. In both experiments, only 20% of our subjects did better in noise with a single ear, roughly similar to the earlier Jerger et al (1993) finding that 8-10% of elderly hearing aid users preferred one hearing aid.

Wind noise in hearing aids with directional and omnidirectional microphones: Polar characteristics of custom-made hearing aids

The purpose of this study was to examine the characteristics of wind noise at the output of in-the-ear, in-the-canal, and completely-in-the-canal hearing aids. The hearing aids were programed to have linear amplification with matching flat frequency responses for directional (DIR) and omnidirectional (OMNI) microphones. The microphone output was then recorded in a quiet wind tunnel when the Knowles electronic manikin for acoustic research (KEMAR) head was turned from 0 degrees to 360 degrees . The overall, 125, 500, and 2000 Hz one-third octave band flow noise levels were calculated and plotted in polar patterns. Correlation coefficients, average differences, and level differences between DIR and OMNI were also calculated. Flow noise levels were the highest when KEMAR was facing the direction of the flow and angles between 190 degrees and 250 degrees . The noise levels were the lowest when the hearing aids were facing the direction of the flow. The polar patterns of DIR and OMNI had similar shapes and DIR generally had higher levels than OMNI. DIR, however, could have lower levels than OMNI in some angles because of its capability to reduce noise in the far field. Comparisons of polar characteristics with behind-the-ear hearing aids, and clinical and engineering design applications of current results are discussed.

한국인 평균 두형 실험더미의 제작과 머리전달함수의 측정

Based on the averaged Korean head shapes that are the results of digital Korean project by KISTI and Catholic university, experimental apparatus of head dummies of Korean male and female are developed in order to measure head-related transfer functions(HRTF) by using a reverse engineering and rapid prototyping techniques. For the Korean dummies, HRTFs are measured using the substitution method ever 12kHz frequency bands. At every azimuth angle <TEX>$15^{\circ}$</TEX> HRTFs are measured for elevation angles <TEX>$-30^{\circ}$</TEX>, <TEX>$0^{\circ}$</TEX> and <TEX>$-30^{\circ}$</TEX>. The measured HRTFs are compared with those of KEMAR(knowles electronic manikin for acoustic research) dummy head, which shows <TEX>$3{\sim}5\;dB$</TEX> difference over <TEX>$4{\sim}5\;kHz$</TEX> kHz frequency band.

Generation of Audible Sound from Two Ultrasonic Beams with Dummy Head

In this study, the nonlinear interaction of two sound beams with a Knowles Electronics manikin for acoustic research (KEMAR) dummy head was investigated experimentally. An audible sound was generated under two conditions: two ultrasonic beams were in parallel and at right angle. It was found that the presence of the dummy head not only increased audible sound levels in KEMAR ears owing to a head scattering effect, but also led to different sound levels in two ears owing to a head shadow effect. Experimental results demonstrated that for the two parallel ultrasonic beams, head effects slightly reduced the audible sound levels behind the dummy head, but increased the audible sound area. Furthermore, it was shown that the audible sound could be generated within the intersection zone of two perpendicular ultrasonic beams, which is useful for localized sound delivery in audio engineering.

Predictive Measures of Directional Benefit Part 2: Verification of Different Approaches to Estimating Directional Benefit

In this investigation, the relation between various directivity measures and subject performance with directional microphone hearing aids was determined. Test devices included first- and second-order directional microphones. Recordings of sentences and noise (Hearing in Noise Test, HINT) were made through each test device in simple, complex, and anisotropic background noise conditions. Twenty-six subjects, with normal hearing, were administered the HINT test recordings and directional benefit was computed. These measures were correlated to theoretical, free-field, and Knowles Electronic Manikin for Acoustic Research (KEMAR) directivity index (DI) values, as well as front-to-back ratios (FBR), in situ signal-to-noise ratios (SNR), and a newly proposed Db SNR, wherein a predictive value of the SNR improvement is calculated as a function of the noise source incidence. The different predictive scores showed high correlation to the measured directional benefit scores in the complex (diffuse-like) background noise condition (r = 0.89 to 0.97, p < 0.05) but not across all background noise conditions (r = 0.45 to 0.97, p < 0.05). The Db SNR approach and the in situ SNR measures provided excellent prediction of subject performance in all background noise conditions (0.85 to 0.97, p < 0.05). None of the predictive measures could account for the effects of reverberation on the speech signal (r = 0.35 to 0.40, p < 0.05). For environments that included a discrete number of noise sources, the in situ SNR and Db SNR estimates were most predictive of subject performance. No predictive approach was indicative of the directional benefit achieved when the speech was also subjected to reverberation (temporal distortion). This finding has implications for real-world estimates of directional benefit.

An Objective Procedure for Evaluation of Adaptive Antifeedback Algorithms in Hearing Aids

This study evaluated the performance of nine adaptive antifeedback algorithms. There were two goals: first, to identify objective procedures that are useful for evaluating these algorithms, and second, to identify strengths and weaknesses of existing algorithms. The algorithms were evaluated in behind-the-ear implementations on the Knowles Electronics Manikin for Acoustic Research (KEMAR). Different acoustic conditions were created by placing a telephone handset or a hat on KEMAR. Electroacoustic techniques were devised to measure the following performance aspects of each algorithm: (1) additional gain made available before oscillation, (2) gain lost in specific frequency regions, (3) reduction of suboscillatory peaks in the frequency response, (4) speed of adaptation to changing acoustic conditions, and (5) robustness in the presence of tonal input signals. For each measurement, performance varied widely across algorithms. No single algorithm was clearly superior or inferior to the others. Generally, the feedback cancellation algorithms were less likely to sacrifice gain in specific frequency regions and better at reducing suboscillatory peaks, whereas the algorithms that used noncancellation techniques were more tolerant of tonal input signals. For those algorithms equipped with special operational modes intended for music listening, the music mode improved the response to tonal inputs but sometimes sacrificed other performance aspects. Algorithms that required an acoustic measurement for initialization purposes tended to perform poorly in acoustic conditions dissimilar to the condition in which initialization was performed. The objective methods devised for this study appear useful for evaluating the performance of adaptive antifeedback algorithms. Currently available algorithms demonstrate a wide range of performance, and further research is required to develop new algorithms that combine the best features of existing algorithms.

Auditory localization of nearby sources. Head-related transfer functions.

Although researchers have long recognized the unique properties of the head-related transfer function (HRTF) for nearby sources (within 1 m of the listener's head), virtually all of the HRTF measurements described in the literature have focused on source locations 1 m or farther from the listener. In this study, HRTFs for sources at distances from 0.12 to 1 m were calculated using a rigid-sphere model of the head and measured using a Knowles Electronic Manikin for Acoustic Research (KEMAR) and an acoustic point source. Both the calculations and the measurements indicate that the interaural level difference (ILD) increases substantially for lateral sources as distance decreases below 1 m, even at low frequencies where the ILD is small for distant sources. In contrast, the interaural time delay (ITD) is roughly independent of distance even when the source is close. The KEMAR measurements indicate that the direction of the source relative to the outer ear plays an important role in determining the high-frequency response of the HRTF in the horizontal plane. However, the elevation-dependent characteristics of the HRTFs are not strongly dependent on distance, and the contribution of the pinna to the HRTF is independent of distance beyond a few centimeters from the ear. Overall, the results suggest that binaural cues play an important role in auditory distance perception for nearby sources.

Personal Cassette Players (‘Walkman’). Do they Cause Noise-Induced Hearing Loss?

Playing selected types of music on five different personal cassette players (PCPs) and using different gain (volume) settings, A-weighted maximum and equivalent sound pressure levels (SPLs) were measured on KEMAR (Knowles Electronics Manikin for Acoustic Research). The octave band SPLs were measured on KEMAR ear and transformed to field values in order to compare measured values with the Norwegian noise risk criteria. Temporary threshold shifts (TTS) measured in 6 subjects after listening to two different pop music cassettes on one PCP in two separate sessions, are presented. Based upon these studies we conclude that the risk of acquiring permanent noise-induced hearing loss (NIHL) from use of PCP is very small for what we found to be normal listening conditions.

Measuring sound attenuation capabilities of helmets using a KEMAR manikin

Sound attenuation measurements were made on three Army helmets and one headset. A Knowles Electronic Manikin for Acoustic Research (KEMAR) was instrumented to read the sound pressure level of one-third octave band filtered white noise. Sound pressure levels were measured using two different procedures. The first procedure employed a Zwislocki coupler microphone responding as a human eardrum and the second procedure used a probe tube located at the ear canal entrance of the manikin. All measurements were taken with the manikin in a diffuse sound field in a sound isolation room. Attenuation values were calculated by subtracting the sound pressure level measured with the helmet on the manikin's head from those sound pressure levels recorded without the helmet in place. These attenuation values compare favorably at most frequency bands with similar measurements made using human subjects in the same sound field wearing the same helmets. [Work supported by U.S. Army, Natick Laboratories, Contract No. DAAK 60-80-C-100.]