Adam A. Hersbach

Improvements in speech perception with an experimental nonlinear frequency compression hearing device.

The performance of an experimental frequency compression hearing device was evaluated using tests of speech understanding in quiet. The device compressed frequencies above a programmable cut-off, resulting in those parts of the input signal being shifted to lower frequencies. Below the cut-off, signals were amplified without frequency shifting. Subjects were experienced hearing aid users with moderate-to-severe sensorineural hearing loss and sloping audiograms. Their recognition of monosyllabic words was tested using the experimental device in comparison with conventional hearing aids. Of the 17 subjects, eight showed a significant score improvement (p < 0.05), whereas one subject showed a significant score decrease. Some of the improvements may have resulted from the better audibility provided in the high frequencies by the experimental device in comparison with the conventional aids. However, a subsequent study found that increasing the high-frequency gain in the conventional aids did not produce equivalent perceptual benefits.

Frequency-compression outcomes in listeners with steeply sloping audiograms

Previous investigation of an experimental, wearable frequency-compression hearing aid revealed improvements in speech perception for a group of listeners with moderately sloping audiograms (Simpson et al, 2005). In the frequency-compression hearing aid, high frequencies (above 1600 Hz) were amplified in addition to being lowered in frequency. Lower frequencies were amplified without frequency shifting. In the present study, an identical frequency-compression scheme was evaluated in a group of seven subjects, all of whom had steeply sloping hearing losses. No significant differences in group mean scores were found between the frequency-compression device and a conventional hearing instrument for understanding speech in quiet. Testing in noise showed improvements for the frequency-compression scheme for only one of the five subjects tested. Subjectively, all but one of the subjects preferred the sound quality of the conventional hearing instruments. In conclusion, the experimental frequency-compression scheme provided only limited benefit to these listeners with steeply sloping hearing losses. Sumario Las investigaciones previas sobre un auxiliar auditivo experimental con compresión de frecuencia revelaron una mejor percepción del lenguaje en un grupo de oyentes con audiogramas de pendiente moderada (Simpson y col., 2005). En el auxiliar auditivo con compresión de frecuencia, las frecuencias agudas (por encima de 1600Hz) se amplificaron además de ser reducidas en frecuencia. Las frecuencias más graves fueron amplificadas sin cambio en la frecuencia. En el presente estudio, un esquema similar de compresión de frecuencia fue evaluado en un grupo de 7 sujetos, todos los cuales tenían hipoacusia con pendientes abruptas. No se encontraron diferencias significativas en los puntajes medios de grupo para entender el lenguaje en silencio entre el dispositivo con compresión de frecuencia y el instrumento convencional. La evaluación en ruido mostró mejoría para el esquema de compresión de la frecuencia en sólo uno de los cinco sujetos evaluados. Subjetivamente, todos menos uno de los sujetos, prefirieron la calidad del sonido de los instrumentos auditivos convencionales. En conclusión, el esquema experimental de compresión de frecuencia aportó sólo un beneficio limitado para los oyentes con hipoacusias de pendiente abrupta.

Clinical Evaluation of Signal-to-Noise Ratio―Based Noise Reduction in Nucleus® Cochlear Implant Recipients

Objective: The aim of this study was to investigate whether a real-time noise reduction algorithm provided speech perception benefit for Cochlear™ Nucleus® cochlear implant recipients in the laboratory. Design: The noise reduction algorithm attenuated masker-dominated channels. It estimated the signal-to-noise ratio of each channel on a short-term basis from a single microphone input, using a recursive minimum statistics method. In this clinical evaluation, the algorithm was implemented in two programs (noise reduction programs 1 [NR1] and 2 [NR2]), which differed in their level of noise reduction. These programs used advanced combination encoder (ACE™) channel selection and were compared with ACE without noise reduction in 13 experienced cochlear implant subjects. An adaptive speech reception threshold (SRT) test provided the signal-to-noise ratio for 50% sentence intelligibility in three different types of noises: speech-weighted, cocktail party, and street-side city noise. Results: In all three noise types, mean SRTs for both NR programs were significantly better than those for ACE. The greatest improvement occurred for speech-weighted noise; the SRT benefit over ACE was 1.77 dB for NR1 and 2.14 dB for NR2. There were no significant differences in speech perception scores between the two NR programs. Subjects reported no degradation in sound quality with the experimental programs. Conclusions: The noise reduction algorithm was successful in improving sentence perception in speech-weighted noise, as well as in more dynamic types of background noise. The algorithm is currently being trialed in a behind-the-ear processor for take-home use.

Combining directional microphone and single-channel noise reduction algorithms: a clinical evaluation in difficult listening conditions with cochlear implant users.

Objectives: This study tested a combination of algorithms designed to improve cochlear implant performance in noise. A noise reduction (NR) algorithm, based on signal to noise ratio estimation was evaluated in combination with several directional microphone algorithms available in the CochlearTM CP810 sound processor. Design: Fourteen adult unilateral cochlear implant users participated in the study. Evaluation was conducted using word recognition in quiet, sentence recognition in noise, and subjective feedback via questionnaire after a period of take-home use. Music appreciation was also evaluated in a controlled listening task. The sentence recognition task measured speech reception threshold for 50% morphemes correct. The interfering maskers were speech-weighted noise and competing talkers, which were spatially separated from the target speech. In addition, the locations of the noise maskers changed during the test in an effort to replicate relevant real-world listening conditions. SmartSoundTM directionality settings Standard, Zoom, and BeamTM (used in the SmartSound programs Everyday, Noise, and Focus, respectively) were all evaluated with and without NR. Results: Microphone directionality demonstrated a consistent benefit in sentence recognition in all noise conditions tested. The group average speech reception threshold benefit over the Standard setting was 3.7 dB for Zoom and 5.3 dB for Beam. Addition of the NR algorithm further improved sentence recognition by 1.3 dB when the noise maskers were speech-weighted noise. There was an overall group preference for the NR algorithm in noisy environments. Group mean word recognition in quiet, preference in quiet conditions, and music appreciation were all unaffected by the NR algorithm. Conclusions: Multimicrophone directionality was effective in improving speech understanding in spatially separated noisy conditions. The single-channel NR algorithm further enhanced speech intelligibility in speech-weighted noise for cochlear implant users while maintaining equivalent performance in quiet situations and when listening to music.

An adaptive Australian Sentence Test in Noise (AuSTIN).

Objectives: The aim of this research is to describe the development of an adaptive Australian Sentence Test in Noise and to validate the test in terms of test–retest reliability and efficiency using data obtained from its clinical application. Design: The relative intelligibility of 1264 Bamford-Kowal-Bench (BKB)–like sentences in the presence of competing four-talker babble was assessed with cochlear implant recipients. Intensity adjustments to the babble segments were made to reduce intersentence variability. Computer software was developed to administer an adaptive speech reception threshold (SRT) test using these adjusted sentence/babble pairs and test–retest SRT data from a separate group of 23 cochlear implant recipients was analyzed, comparing different SRT calculation and test stopping rules. Results: The adjusted sentence/babble pairs were used in clinical studies to obtain an SRT by presenting 32 sentences. Analysis of test–retest pairs of SRT data from 23 recipients indicated that a psychometric fit SRT calculation rule provided better reliability than did the Hearing in Noise Test (HINT) calculation rule, or rules based on mean turns. This rule, using the morpheme correct scores for each sentence, gave a standard deviation for a single SRT of 0.76 dB. Further analyses revealed that the test could be shortened to 20 sentences with an increase of 0.19 dB in variability, while reducing the median test time by approximately 2 min. Conclusions: This article reports validation data for a new Australian Sentence Test In Noise. When 20 BKB–like sentences are used with a psychometric fit calculation rule, a standard deviation of approximately 1 dB is obtained in approximately 3 min 36 sec.

A beamformer post-filter for cochlear implant noise reduction

Cochlear implant users have limited ability to understand speech in noisy conditions. Signal processing methods to address this issue that use multiple microphones typically use beamforming to perform noise reduction. However, the effectiveness of the beamformer is diminished as the number of interfering noises increases and the acoustic environment becomes more diffuse. A multi-microphone noise reduction algorithm that aims to address this issue is presented in this study. The algorithm uses spatial filtering to estimate the signal-to-noise ratio (SNR) and attenuates time-frequency elements that have poor SNR. The algorithm was evaluated by measuring intelligibility of speech embedded in 4-talker babble where the interfering talkers were spatially separated and changed location during the test. Twelve cochlear implant users took part in the evaluation, which demonstrated a significant mean improvement of 4.6 dB (standard error 0.4, P < 0.001) in speech reception threshold compared to an adaptive beamformer. The results suggest that a substantial improvement in performance can be gained for cochlear implant users in noisy environments where the noise is spatially separated from the target speech.

Algorithms to improve listening in noise for cochlear implant users

In this study, a two-microphone noise reduction algorithm that aims to improve directional beamformer performance for cochlear implant (CI) users is presented. The algorithm is computationally inexpensive and estimates a spatially-based signal-to-noise ratio (SNR), attenuating time-frequency elements that have poor SNR. The attenuation function was specifically tuned for application to CI. Using a real-time implementation of the algorithm, evaluation took place in a variety of noisy situations in which the competing speech sources were spatially separated and changed location during the test. Speech intelligibility tests with CI users revealed the new algorithm improved speech performance by 4.6 dB in speech reception threshold (SRT) compared to a commercially available adaptive beamformer.

A Wavelet-Based Noise Reduction Algorithm and Its Clinical Evaluation in Cochlear Implants

Noise reduction is often essential for cochlear implant (CI) recipients to achieve acceptable speech perception in noisy environments. Most noise reduction algorithms applied to audio signals are based on time-frequency representations of the input, such as the Fourier transform. Algorithms based on other representations may also be able to provide comparable or improved speech perception and listening quality improvements. In this paper, a noise reduction algorithm for CI sound processing is proposed based on the wavelet transform. The algorithm uses a dual-tree complex discrete wavelet transform followed by shrinkage of the wavelet coefficients based on a statistical estimation of the variance of the noise. The proposed noise reduction algorithm was evaluated by comparing its performance to those of many existing wavelet-based algorithms. The speech transmission index (STI) of the proposed algorithm is significantly better than other tested algorithms for the speech-weighted noise of different levels of signal to noise ratio. The effectiveness of the proposed system was clinically evaluated with CI recipients. A significant improvement in speech perception of 1.9 dB was found on average in speech weighted noise.

Perceptual effect of reverberation on multi-microphone noise reduction for cochlear implants

The combination of noise and reverberation make listening conditions difficult for cochlear implant (CI) users. The perceptual effect of reverberation was evaluated via speech intelligibility tests with CI users. A fixed directional microphone, an adaptive directional microphone and a beamformer post-filter were evaluated. Reverberation was varied by changing the target and noise distance and by simulating a highly reverberant room with concrete surfaces. CI performance expectedly degraded as the target distance was increased, but the benefit of noise reduction was unaffected by listening distance. In the highly reverberant condition, CI performance was severely degraded, but noise reduction benefit remarkably increased, especially for the beamformer post-filter algorithm. All directional processing algorithms were suitable for use in noisy reverberant conditions and the best outcome was provided by the post-filter condition.