RESUMEN
Perception adapts to the properties of prior stimulation, as illustrated by phenomena such as visual color constancy or speech context effects. In the auditory domain, only little is known about adaptive processes when it comes to the attribute of auditory brightness. Here, we report an experiment that tests whether listeners adapt to spectral colorations imposed on naturalistic music and speech excerpts. Our results indicate consistent contrastive adaptation of auditory brightness judgments on a trial-by-trial basis. The pattern of results suggests that these effects tend to grow with an increase in the duration of the adaptor context but level off after around 8 trials of 2 s duration. A simple model of the response criterion yields a correlation of r = .97 with the measured data and corroborates the notion that brightness perception adapts on timescales that fall in the range of auditory short-term memory. Effects turn out to be similar for spectral filtering based on linear spectral filter slopes and filtering based on a measured transfer function from a commercially available hearing device. Overall, our findings demonstrate the adaptivity of auditory brightness perception under realistic acoustical conditions.
Asunto(s)
Estimulación Acústica , Adaptación Fisiológica , Percepción Auditiva/fisiología , Umbral Auditivo , Audición/fisiología , Música , Percepción del Habla/fisiología , Adulto , Femenino , Humanos , Masculino , Adulto JovenRESUMEN
Smart headphones or hearables use different types of algorithms such as noise cancelation, feedback suppression, and sound pressure equalization to eliminate undesired sound sources or to achieve acoustical transparency. Such signal processing strategies might alter the spectral composition or interaural differences of the original sound, which might be perceived by listeners as monaural or binaural distortions and thus degrade audio quality. To evaluate the perceptual impact of these distortions, subjective quality ratings can be used, but these are time consuming and costly. Auditory-inspired instrumental quality measures can be applied with less effort and may also be helpful in identifying whether the distortions impair the auditory representation of monaural or binaural cues. Therefore, the goals of this study were (a) to assess the applicability of various monaural and binaural audio quality models to distortions typically occurring in hearables and (b) to examine the effect of those distortions on the auditory representation of spectral, temporal, and binaural cues. Results showed that the signal processing algorithms considered in this study mainly impaired (monaural) spectral cues. Consequently, monaural audio quality models that capture spectral distortions achieved the best prediction performance. A recent audio quality model that predicts monaural and binaural aspects of quality was revised based on parts of the current data involving binaural audio quality aspects, leading to improved overall performance indicated by a mean Pearson linear correlation of 0.89 between obtained and predicted ratings.
Asunto(s)
Señales (Psicología) , Localización de Sonidos , Estimulación Acústica , Algoritmos , Humanos , Ruido , TecnologíaRESUMEN
Acoustic feedback in hearing aids occurs due to the coupling between the hearing aid loudspeaker and microphones. In order to reduce acoustic feedback, adaptive filters are often used to estimate the feedback path. To increase the convergence speed and decrease the computational complexity of the adaptive algorithms, it has been proposed to split the acoustic feedback path into a time-invariant fixed part and a time-varying variable part. A key question of this approach is how to determine the fixed part. In this paper, two approaches are investigated: (1) a digital filter design approach that makes use of the signals of at least two hearing aid microphones and (2) a defined physical location approach using an electro-acoustic model and the signals of one hearing aid microphone and an additional ear canal microphone. An experimental comparison using measured acoustic feedback paths showed that both approaches enable one to reduce the number of variable part coefficients. It is shown that individualization of the fixed part increases the performance. Furthermore, the two approaches offer solutions for different requirements on the effort to a specific hearing aid design on the one hand and the effort during the hearing aid fitting on the other hand.
Asunto(s)
Audífonos , Procesamiento de Señales Asistido por Computador , Estimulación Acústica , Acústica , Diseño de Equipo , Retroalimentación , Humanos , Modelos Biológicos , Modelos TeóricosRESUMEN
Adaptive feedback cancellation (AFC) techniques are common in modern hearing aid devices (HADs) since these techniques have been successful in increasing the stable gain. Accordingly, there has been a significant effort to improve AFC technology, especially for open-fitting and in-ear HADs, for which howling is more prevalent due to the large acoustic coupling between the loudspeaker and the microphone. In this paper, the authors propose a hybrid AFC (H-AFC) scheme that is able to shorten the time it takes to recover from howling. The proposed H-AFC scheme consists of a switched combination adaptive filter, which is controlled by a soft-clipping-based stability detector to select either the standard normalized least mean squares (NLMS) algorithm or the prediction-error-method (PEM) NLMS algorithm to update the adaptive filter. The standard NLMS algorithm is used to obtain fast convergence, while the PEM-NLMS algorithm is used to provide a low bias solution. This stability-controlled adaptation is hence the means to improve performance in terms of both convergence rate as well as misalignment, while only slightly increasing computational complexity. The proposed H-AFC scheme has been evaluated for both speech and music signals, resulting in a significantly improved convergence and re-convergence rate, i.e., a shorter howling period, as well as a lower average misalignment and a larger added stable gain compared to using either the NLMS or the PEM-NLMS algorithm alone. An objective evaluation using the perceptual evaluation of speech quality and the perceptual evaluation of audio quality measures shows that the proposed H-AFC scheme provides very high-quality speech and music signals. This has also been verified through a subjective listening experiment with N = 15 normal-hearing subjects using a multi-stimulus test with hidden reference and anchor, showing that the proposed H-AFC scheme results in a better perceptual quality than the state-of-the-art PEM-NLMS algorithm.
Asunto(s)
Acústica , Algoritmos , Percepción Auditiva , Corrección de Deficiencia Auditiva/instrumentación , Audífonos , Personas con Deficiencia Auditiva/rehabilitación , Procesamiento de Señales Asistido por Computador , Estimulación Acústica , Adulto , Diseño de Equipo , Humanos , Modelos Teóricos , Música , Personas con Deficiencia Auditiva/psicología , Espectrografía del Sonido , Inteligibilidad del Habla , Percepción del HablaRESUMEN
In many applications in which speech is played back via a sound reinforcement system such as public address systems and mobile phones, speech intelligibility is degraded by additive environmental noise. A possible solution to maintain high intelligibility in noise is to pre-process the speech signal based on the estimated noise power at the position of the listener. The previously proposed AdaptDRC algorithm [Schepker, Rennies, and Doclo (2015). J. Acoust. Soc. Am. 138, 2692-2706] applies both frequency shaping and dynamic range compression under an equal-power constraint, where the processing is adaptively controlled by short-term estimates of the speech intelligibility index. Previous evaluations of the algorithm have focused on normal-hearing listeners. In this study, the algorithm was extended with an adaptive gain stage under an equal-peak-power constraint, and evaluated with eleven normal-hearing and ten mildly to moderately hearing-impaired listeners. For normal-hearing listeners, average improvements in speech reception thresholds of about 4 and 8 dB compared to the unprocessed reference condition were measured for the original algorithm and its extension, respectively. For hearing-impaired listeners, the average improvements were about 2 and 6 dB, indicating that the relative improvement due to the proposed adaptive gain stage was larger for these listeners than the benefit of the original processing stages.
Asunto(s)
Acústica , Algoritmos , Ruido/efectos adversos , Enmascaramiento Perceptual , Personas con Deficiencia Auditiva/psicología , Presbiacusia/psicología , Procesamiento de Señales Asistido por Computador , Inteligibilidad del Habla , Percepción del Habla , Estimulación Acústica , Adulto , Anciano , Audiometría de Tonos Puros , Umbral Auditivo , Estudios de Casos y Controles , Femenino , Audición , Humanos , Masculino , Persona de Mediana Edad , Presbiacusia/diagnóstico , Presbiacusia/fisiopatología , Prueba del Umbral de Recepción del Habla , Adulto JovenRESUMEN
OBJECTIVE: The purpose of this study was to assess perceived listening effort and speech intelligibility in reverberant and noisy conditions for hearing-impaired listeners for conditions that are similar according to the speech transmission index (STI). DESIGN: Scaled listening effort was measured in four different conditions at five different STI generated using various relative contributions of noise and reverberant interferences. Intelligibility was measured for a subset of conditions. STUDY SAMPLE: Twenty mildly to moderately hearing-impaired listeners. RESULTS: In general, listening effort decreased and speech intelligibility increased with increasing STI. For simulated impulse responses consisting of white Gaussian noise exponentially decaying in time, a good agreement between conditions of different relative contributions of noise and reverberation was found. For real impulse responses, the STI slightly overestimated the effect of reverberation on the perceived listening effort and underestimated its effect on speech intelligibility. Including the average hearing loss in the calculation of the STI led to a better agreement between STI predictions and subjective data. CONCLUSION: Speech intelligibility and listening effort provide complementary tools to evaluate speech perception over a broad range of acoustic scenarios. In addition, when incorporating hearing loss information the STI provides a rough prediction of listening effort in these acoustic scenarios.
Asunto(s)
Estimulación Acústica/métodos , Pérdida Auditiva Sensorineural/psicología , Ruido , Inteligibilidad del Habla , Anciano , Anciano de 80 o más Años , Umbral Auditivo , Femenino , Humanos , Masculino , Persona de Mediana Edad , Percepción del HablaRESUMEN
In many speech communication applications, such as public address systems, speech is degraded by additive noise, leading to reduced speech intelligibility. In this paper a pre-processing algorithm is proposed that is capable of increasing speech intelligibility under an equal-power constraint. The proposed AdaptDRC algorithm comprises two time- and frequency-dependent stages, i.e., an amplification stage and a dynamic range compression stage that are both dependent on the Speech Intelligibility Index (SII). Experiments using two objective measures, namely, the extended SII and the short-time objective intelligibility measure (STOI), and a formal listening test were conducted to compare the AdaptDRC algorithm with a modified version of a recently proposed algorithm in three different noise conditions (stationary car noise and speech-shaped noise and non-stationary cafeteria noise). While the objective measures indicate a similar performance for both algorithms, results from the formal listening test indicate that for the two stationary noises both algorithms lead to statistically significant improvements in speech intelligibility and for the non-stationary cafeteria noise only the proposed AdaptDRC algorithm leads to statistically significant improvements. A comparison of both objective measures and results from the listening test shows high correlations, although, in general, the performance of both algorithms is overestimated.
Asunto(s)
Algoritmos , Inteligibilidad del Habla , Percepción del Habla/fisiología , Adulto , Compresión de Datos , Femenino , Humanos , Masculino , Relación Señal-Ruido , Adulto JovenRESUMEN
A reciprocal measurement procedure to measure the acoustic feedback path in hearing aids is investigated. The advantage of the reciprocal measurement compared to the direct measurement is a significantly reduced sound pressure in the ear. The direct and reciprocal measurements are compared using measurements on a dummy head with adjustable ear canals, different earmolds, and variations in the outer sound field. The results show that the reciprocal measurement procedure can be used to obtain plausible feedback paths, while reducing the sound pressure in the ear canal by 30 to 40 dB.
Asunto(s)
Retroalimentación , Audífonos , Estimulación Acústica , Conducto Auditivo Externo , Diseño de Equipo , Humanos , Modelos Anatómicos , Ruido , Presión , Acústica del Lenguaje , Transductores de PresiónRESUMEN
This study compared the combined effect of noise and reverberation on listening effort and speech intelligibility to predictions of the speech transmission index (STI). Listening effort was measured in normal-hearing subjects using a scaling procedure. Speech intelligibility scores were measured in the same subjects and conditions: (a) Speech-shaped noise as the only interfering factor, (b) + (c) fixed signal-to-noise ratios (SNRs) of 0 or 7 dB and reverberation as detrimental factors, and (d) reverberation as the only detrimental factor. In each condition, SNR and reverberation were combined to produce STI values of 0.17, 0.30, 0.43, 0.57, and 0.70, respectively. Listening effort always decreased with increasing STI, thus enabling a rough prediction, but a significant bias was observed indicating that listening effort was lower in reverberation only than in noise only at the same STI for one type of impulse responses. Accordingly, speech intelligibility increased with increasing STI and was significantly better in reverberation only than in noise only at the same STI. Further analyses showed that the broadband reverberation time is not always a good estimate of speech degradation in reverberation and that different speech materials may differ in their robustness toward detrimental effects of reverberation.