scholarly journals Tracking of Noise Tolerance to Predict Hearing Aid Satisfaction in Loud Noisy Environments

2019 ◽  
Vol 30 (04) ◽  
pp. 302-314
Author(s):  
Eric Seper ◽  
Francis Kuk ◽  
Petri Korhonen ◽  
Christopher Slugocki

AbstractA method that tracked tolerable noise level (TNL) over time while maintaining subjective speech intelligibility was reported previously. Although this method was reliable and efficacious as a research tool, its clinical efficacy and predictive ability of real-life hearing aid satisfaction were not measured.The study evaluated an adaptive method to estimate TNL using slope and variance of tracked noise level as criteria in a clinical setting. The relationship between TNL and subjective hearing aid satisfaction in noisy environments was also investigated.A single-blinded, repeated-measures design.Seventeen experienced hearing aid wearers with bilateral mild-to-moderately-severe sensorineural hearing loss.Participants listened to 82-dB SPL continuous speech and tracked the background noise level that they could “put up with” while subjectively understanding >90% of the speech material. Two trials with each babble noise and continuous speech-shaped noise were measured in a single session. All four trials were completed aided using the participants’ own hearing aids. The stimuli were presented in the sound field with speech from 0° and noise from the 180° azimuth. The instantaneous tolerable noise level was measured using a custom program and scored in two ways; the averaged TNL (aTNL) over the 2-min trial and the estimated TNL (eTNL) as soon as the listeners reached a stable noise estimate. Correlation between TNL and proportion of satisfied noisy environments was examined using the MarkeTrak questionnaire.All listeners completed the tracking of noise tolerance procedure within 2 min with good reliability. Sixty-five percent of the listeners yielded a stable noise estimate after 59.9 sec of actual test time. The eTNL for all trials was 78.6 dB SPL (standard deviation [SD] = 4.4 dB). The aTNL for all trials was 78.0 dB SPL (SD = 3.3 dB) after 120 sec. The aTNL was 79.2 dB SPL (SD = 5.4 dB) for babble noise and 77.0 dB SPL (SD = 5.9 dB) for speech-shaped noise. High within-session test–retest reliability was evident. The 95% confidence interval was 1.5 dB for babble noise and 2.8 dB for continuous speech-shaped noise. No significant correlation was measured between overall hearing aid satisfaction and the aTNL (ρ = 0.20 for both noises); however, a significant relationship between aTNL and proportion of satisfied noisy situations was evident (ρ = 0.48 for babble noise and ρ = 0.55 for speech-shaped noise).The eTNL scoring method yielded similar results as the aTNL method although requiring only half the time for 65% of the listeners. This time efficiency, along with its reliability and the potential relationship between TNL and hearing aid satisfaction in noisy listening situations suggests that this procedure may be a good clinical tool to evaluate whether specific features on a hearing aid would improve noise tolerance and predict wearer satisfaction with the selected hearing aid in real-life loud noisy situations. A larger sample of hearing aid wearers is needed to further validate these potential uses.

2017 ◽  
Vol 28 (08) ◽  
pp. 698-707
Author(s):  
Francis Kuk ◽  
Eric Seper ◽  
Chi-Chuen Lau ◽  
Petri Korhonen

AbstractThe benefits offered by noise reduction (NR) features on a hearing aid had been studied traditionally using test conditions that set the hearing aids into a stable state of performance. While adequate, this approach does not allow the differentiation of two NR algorithms that differ in their timing characteristics (i.e., activation and stabilization time).The current study investigated a new method of measuring noise tolerance (Tracking of Noise Tolerance [TNT]) as a means to differentiate hearing aid technologies. The study determined the within-session and between-session reliability of the procedure. The benefits provided by various hearing aid conditions (aided, two NR algorithms, and a directional microphone algorithm) were measured using this procedure. Performance on normal-hearing listeners was also measured for referencing.A single-blinded, repeated-measures design was used.Thirteen experienced hearing aid wearers with a bilaterally symmetrical (≤10 dB) mild-to-moderate sensorineural hearing loss participated in the study. In addition, seven normal-hearing listeners were tested in the unaided condition.Participants tracked the noise level that met the criterion of tolerable noise level (TNL) in the presence of an 85 dB SPL continuous discourse passage. The test conditions included an unaided condition and an aided condition with combinations of NR and microphone modes within the UNIQUE hearing aid (omnidirectional microphone, no NR; omnidirectional microphone, NR; directional microphone, no NR; and directional microphone, NR) and the DREAM hearing aid (omnidirectional microphone, no NR; omnidirectional microphone, NR). Each tracking trial lasted 2 min for each hearing aid condition. Normal-hearing listeners tracked in the unaided condition only. Nine of the 13 hearing-impaired listeners returned after 3 mo for retesting in the unaided and aided conditions with the UNIQUE hearing aid. The individual TNL was estimated for each participant for all test conditions. The TNT index was calculated as the difference between 85 dB SPL and the TNL.The TNT index varied from 2.2 dB in the omnidirectional microphone, no NR condition to −4.4 dB in the directional microphone, NR on condition. Normal-hearing listeners reported a TNT index of −5.7 dB using this procedure. The averaged improvement in TNT offered by the NR algorithm on the UNIQUE varied from 2.1 dB when used with a directional microphone to 3.0 dB when used with the omnidirectional microphone. The time course of the NR algorithm was different between the UNIQUE and the DREAM hearing aids, with the UNIQUE reaching a stable TNL sooner than the DREAM. The averaged improvement in TNT index from the UNIQUE directional microphone was 3.6 dB when NR was activated and 4.4 dB when NR was deactivated. Together, directional microphone and NR resulted in a total TNT improvement of 6.5 dB. The test–retest reliability of the procedure was high, with an intrasession 95% confidence interval (CI) of 2.2 dB and an intersession 95% CI of 4.2 dB.The effect of the NR and directional microphone algorithms was measured to be 2–3 and 3.6–4.4 dB, respectively, using the TNT procedure. Because of its tracking property and reliability, this procedure may hold promise in differentiating among some hearing aid features that also differ in their time course of action.


2017 ◽  
Vol 28 (01) ◽  
pp. 046-057 ◽  
Author(s):  
Petri Korhonen ◽  
Francis Kuk ◽  
Eric Seper ◽  
Martin Mørkebjerg ◽  
Majken Roikjer

AbstractWind noise is a common problem reported by hearing aid wearers. The MarkeTrak VIII reported that 42% of hearing aid wearers are not satisfied with the performance of their hearing aids in situations where wind is present.The current study investigated the effect of a new wind noise attenuation (WNA) algorithm on subjective annoyance and speech recognition in the presence of wind.A single-blinded, repeated measures design was used.Fifteen experienced hearing aid wearers with bilaterally symmetrical (≤10 dB) mild-to-moderate sensorineural hearing loss participated in the study.Subjective rating for wind noise annoyance was measured for wind presented alone from 0° and 290° at wind speeds of 4, 5, 6, 7, and 10 m/sec. Phoneme identification performance was measured using Widex Office of Clinical Amplification Nonsense Syllable Test presented at 60, 65, 70, and 75 dB SPL from 270° in the presence of wind originating from 0° at a speed of 5 m/sec.The subjective annoyance from wind noise was reduced for wind originating from 0° at wind speeds from 4 to 7 m/sec. The largest improvement in phoneme identification with the WNA algorithm was 48.2% when speech was presented from 270° at 65 dB SPL and the wind originated from 0° azimuth at 5 m/sec.The WNA algorithm used in this study reduced subjective annoyance for wind speeds ranging from 4 to 7 m/sec. The algorithm was effective in improving speech identification in the presence of wind originating from 0° at 5 m/sec. These results suggest that the WNA algorithm used in the current study could expand the range of real-life situations where a hearing-impaired person can use the hearing aid optimally.


2015 ◽  
Vol 26 (05) ◽  
pp. 478-493 ◽  
Author(s):  
Francis Kuk ◽  
Eric Seper ◽  
Chi Lau ◽  
Bryan Crose ◽  
Petri Korhonen

Background: Bilateral contralateral routing of signals (BiCROS) hearing aids function to restore audibility of sounds originating from the side of the unaidable ear. However, when speech is presented to the side of the aidable ear and noise to the side of the unaidable ear, a BiCROS arrangement may reduce intelligibility of the speech signal. This negative effect may be circumvented if an on/off switch is available on the contralateral routing of signals (CROS) transmitter. Purpose: This study evaluated if the proper use of the on/off switch on a CROS transmitter could enhance speech recognition in noise and sound localization abilities. The participants’ subjective reactions to the use of the BiCROS, including the use of the on/off switch in real-life were also evaluated. Research Design: A between-subjects, repeated-measures design was used to assess differences in speech recognition (in quiet and in noise) and localization abilities under four hearing aid conditions (unaided, unilaterally aided, fixed BiCROS setting, and adjusted BiCROS setting) with speech and noise stimuli presented from different azimuths. Participants were trained on the use of the on/off switch on the BiCROS transmitter before testing in the adjusted BiCROS settings. Subjective ratings were obtained with the Speech, Spatial, and Sound Quality (SSQ) questionnaire and a custom questionnaire. Study Sample: Nine adult BiCROS candidates participated in this study. Data Collection and Analysis: Participants wore the Widex Dream-m-CB hearing aid on the aidable ear for 1 week. They then wore the BiCROS for the remainder of the study. Speech recognition and localization testing were completed in four hearing aid conditions (unaided, unilateral aided, fixed BiCROS, and adjusted BiCROS). Speech recognition was evaluated during the first three visits, whereas localization was evaluated over the course of the study. Participants completed the SSQ questionnaire before each visit. The CROS questionnaire was completed at the final visit. A repeated measures analysis of variance with Bonferroni post hoc analysis was used to evaluate the significance of the results on speech recognition, localization, and the SSQ. Results: The results revealed that the adjusted BiCROS condition improved speech recognition scores by 20 rau (rationalized arcsine unit) when speech was presented to the aidable ear and localization by 37% when sounds are presented from the side of the unaidable ear over the fixed BiCROS condition. Statistically significant benefit on the SSQ was also noted with the adjusted BiCROS condition compared to the unilateral fitting. Conclusions: These findings supported the value of an on/off switch on a CROS transmitter because it allows convenient selective transmission of sounds. It also highlighted the importance of instructions and practice in using the BiCROS hearing aid successfully.


2010 ◽  
Vol 21 (04) ◽  
pp. 249-266 ◽  
Author(s):  
Lynzee N. Alworth ◽  
Patrick N. Plyler ◽  
Monika Bertges Reber ◽  
Patti M. Johnstone

Background: Open canal hearing instruments differ in method of sound delivery to the ear canal, distance between the microphone and the receiver, and physical size of the devices. Moreover, RITA (receiver in the aid) and RITE (receiver in the ear) hearing instruments may also differ in terms of retention and comfort as well as ease of use and care for certain individuals. What remains unclear, however, is if any or all of the abovementioned factors contribute to hearing aid outcome. Purpose: To determine the effect of receiver location on performance and/or preference of listeners using open canal hearing instruments. Research Design: An experimental study in which subjects were exposed to a repeated measures design. Study Sample: Twenty-five adult listeners with mild sloping to moderately severe sensorineural hearing loss (mean age 67 yr). Data Collection and Analysis: Participants completed two six-week trial periods for each device type. Probe microphone, objective, and subjective measures (quiet, noise) were conducted unaided and aided at the end of each trial period. Results: Occlusion effect results were not significantly different between the RITA and RITE instruments; however, frequency range was extended in the RITE instruments, resulting in significantly greater maximum gain for the RITE instruments than the RITA instruments at 4000 and 6000 Hz. Objective performance in quiet or in noise was unaffected by receiver location. Subjective measures revealed significantly greater satisfaction ratings for the RITE than for the RITA instruments. Similarly, preference in quiet and overall preference were significantly greater for the RITE than for the RITA instruments. Conclusions: Although no occlusion differences were noted between instruments, the RITE did demonstrate a significant difference in reserve gain before feedback at 4000 and 6000 Hz. Objectively; no positive benefit was noted between unaided and aided conditions on speech recognition tests. These results suggest that such testing may not be sensitive enough to determine aided benefit with open canal instruments. However, the subjective measures (Abbreviated Profile of Hearing Aid Benefit [APHAB] and subjective ratings) did indicate aided benefit for both instruments when compared to unaided. This further suggests the clinical importance of subjective measures as a way to measure aided benefit of open-fit devices.


2018 ◽  
Vol 29 (04) ◽  
pp. 273-278
Author(s):  
Haihong Liu ◽  
Yuanhu Liu ◽  
Ying Li ◽  
Xin Jin ◽  
Jing Li ◽  
...  

AbstractWide dynamic range compression (WDRC) has been widely used in hearing aid technology. However, several reports indicate that WDRC may improve audibility at the expense of speech intelligibility. As such, a modified amplification compression scheme, named adaptive compression, was developed. However, the effect of compression strategies on speech perception in pediatric hearing aid users has not been clearly reported.The purpose of the present study was to investigate the effect of adaptive compression and fast-acting WDRC processing strategies on sentence recognition in noise with Mandarin, pediatric hearing aid users.This study was set up using a double-blind, within-subject, repeated-measures design.Twenty-six children who spoke Mandarin Chinese as their primary language and had bilateral sensorineural hearing loss participated in the study.Sentence recognition in noise was evaluated in behind-the-ear technology with both adaptive compression processing and fast-acting WDRC processing and was selected randomly for each child. Percent correct sentence recognition in noise with fast-acting WDRC and adaptive compression was collected from each participant. Correlation analysis was performed to examine the effect of gender, age at assessment, and hearing threshold of the better ear on signal-to-noise ratio, and a paired-samples t test was employed to compare the performance of the adaptive compression strategy and fast-acting WDRC processing.The mean percentage correct of sentence recognition in noise with behind-the-ear technology with fast-acting WDRC and adaptive compression processing were 62.24% and 68.71%, respectively. The paired-samples t test showed that the performance of the adaptive compression strategy was significantly better than the fast-acting WDRC processing (t = 3.190, p = 0.004).Compared with the fast-acting WDRC, adaptive compression provided better sentence recognition in noise for Mandarin pediatric hearing aid users.


2018 ◽  
Vol 23 (1) ◽  
pp. 32-38 ◽  
Author(s):  
Jantien L. Vroegop ◽  
Nienke C. Homans ◽  
André Goedegebure ◽  
J. Gertjan Dingemanse ◽  
Teun van Immerzeel ◽  
...  

Although the benefit of bimodal listening in cochlear implant users has been agreed on, speech comprehension remains a challenge in acoustically complex real-life environments due to reverberation and disturbing background noises. One way to additionally improve bimodal auditory performance is the use of directional microphones. The objective of this study was to investigate the effect of a binaural beamformer for bimodal cochlear implant (CI) users. This prospective study measured speech reception thresholds (SRT) in noise in a repeated-measures design that varied in listening modality for static and dynamic listening conditions. A significant improvement in SRT of 4.7 dB was found with the binaural beamformer switched on in the bimodal static listening condition. No significant improvement was found in the dynamic listening condition. We conclude that there is a clear additional advantage of the binaural beamformer in bimodal CI users for predictable/static listening conditions with frontal target speech and spatially separated noise sources.


2019 ◽  
Vol 9 (2) ◽  
pp. 17 ◽  
Author(s):  
Roberto Giorgio Rizzo ◽  
Andrea Calimera

Adaptive Voltage Over-Scaling can be applied at run-time to reach the best tradeoff between quality of results and energy consumption. This strategy encompasses the concept of timing speculation through some level of approximation. How and on which part of the circuit to implement such approximation is an open issue. This work introduces a quantitative comparison between two complementary strategies: Algorithmic Noise Tolerance and Approximate Error Detection. The first implements a timing speculation by means approximate computing, while the latter exploits a more sophisticated approach that is based on the approximation of the error detection mechanism. The aim of this study was to provide both a qualitative and quantitative analysis on two real-life digital circuits mapped onto a state-of-the-art 28-nm CMOS technology.


Micromachines ◽  
2019 ◽  
Vol 10 (12) ◽  
pp. 885
Author(s):  
Shuo Zhang ◽  
Ruiqing Zhang ◽  
Shijie Chang ◽  
Chengyu Liu ◽  
Xianzheng Sha

Along with the great performance in diagnosing cardiovascular diseases, current stethoscopes perform unsatisfactorily in controlling undesired noise caused by the surrounding environment and detector operation. In this case, a low-noise-level heart sound system was designed to inhibit noise by a novel thorax-integration head with a flexible electric film. A hardware filter bank and wavelet-based algorithm were employed to enhance the recorded heart sounds from the system. In the experiments, we used the new system and the 3M™ Littmann® Model 3200 Electronic Stethoscope separately to record heart sounds in different noisy environments. The results illustrated that the average estimated noise ratio represented 21.26% and the lowest represented only 12.47% compared to the 3M stethoscope, demonstrating the better performance in denoising ability of this system than state-of-the-art equipment. Furthermore, based on the heart sounds recorded with this system, some diagnosis results were achieved from an expert and compared to echocardiography reports. The diagnoses were correct except for two uncertain items, which greatly confirmed the fact that this system could reserve complete pathological information in the end.


Geophysics ◽  
2020 ◽  
Vol 85 (6) ◽  
pp. KS197-KS206
Author(s):  
Dmitry Alexandrov ◽  
Leo Eisner ◽  
Umair bin Waheed ◽  
SanLinn I. Kaka ◽  
Stewart Alan Greenhalgh

Microseismic monitoring aims at detecting as weak events as possible and providing reliable locations and source mechanisms for these events. Surface monitoring arrays suffer from significant variations of noise levels across receiver lines. When using a large monitoring array, we use a stacking technique to detect microseismic events through maximizing the signal-to-noise ratio (S/N) of the stack. But some receivers with a high noise level do not contribute to improving the S/N of the stack. We have derived a theoretical concept for the proper selection of receivers that best contribute to the stack for a constant strength of a signal across the array. This receiver selection criterion, based on the assumption of constant signal amplitude, provides a robust estimate of the noise threshold level, which could be used to discard or suppress contribution from the receivers that do not improve the S/N of the stack. We found that limiting the number of receivers for stacking improves the location accuracy and reduces the computational cost of data processing. Although the assumption of a constant signal never holds in real-life seismic applications, the noise level varies across the surface receivers in a significantly wider range than the signal amplitude. These noise variations can also increase the uncertainty of the source mechanism inversion and should be accounted for. Synthetic and field data examples show that weighted least-squares inversion with receiver weighting according to the noise level produces more accurate estimates for source mechanisms compared to the inversion that ignores information about noise.


Sensors ◽  
2020 ◽  
Vol 20 (15) ◽  
pp. 4117
Author(s):  
V. D. Ambeth Kumar ◽  
S. Malathi ◽  
Abhishek Kumar ◽  
Prakash M ◽  
Kalyana C. Veluvolu

To communicate efficiently with a prospective user, auditory interfaces are employed in mobile communication devices. Diverse sounds in different volumes are used to alert the user in various devices such as mobile phones, modern laptops and domestic appliances. These alert noises behave erroneously in dynamic noise environments, leading to major annoyances to the user. In noisy environments, as sounds can be played quietly, this leads to the improper masked rendering of the necessary information. To overcome these issues, a multi-model sensing technique is developed as a smartphone application to achieve automatic volume control in a smart phone. Based on the ambient environment, the volume is automatically controlled such that it is maintained at an appropriate level for the user. By identifying the average noise level of the ambient environment from dynamic microphone and together with the activity recognition data obtained from the inertial sensors, the automatic volume control is achieved. Experiments are conducted with five different mobile devices at various noise-level environments and different user activity states. Results demonstrate the effectiveness of the proposed application for active volume control in dynamic environments.


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