Uwe Rass

Signal processing in high-end hearing aids: state of the art, challenges, and future trends

The development of hearing aids incorporates two aspects, namely, the audiological and the technical point of view. The former focuses on items like the recruitment phenomenon, the speech intelligibility of hearing-impaired persons, or just on the question of hearing comfort. Concerning these subjects, different algorithms intending to improve the hearing ability are presented in this paper. These are automatic gain controls, directional microphones, and noise reduction algorithms. Besides the audiological point of view, there are several purely technical problems which have to be solved. An important one is the acoustic feedback. Another instance is the proper automatic control of all hearing aid components by means of a classification unit. In addition to an overview of state-of-the-art algorithms, this paper focuses on future trends.

The effect of multi-channel wide dynamic range compression, noise reduction, and the directional microphone on horizontal localization performance in hearing aid wearers

This study examined the effect that signal processing strategies used in modern hearing aids, such as multi-channel WDRC, noise reduction, and directional microphones have on interaural difference cues and horizontal localization performance relative to linear, time-invariant amplification. Twelve participants were bilaterally fitted with BTE devices. Horizontal localization testing using a 360° loudspeaker array and broadband pulsed pink noise was performed two weeks, and two months, post-fitting. The effect of noise reduction was measured with a constant noise present at 80° azimuth. Data were analysed independently in the left/right and front/back dimension and showed that of the three signal processing strategies, directional microphones had the most significant effect on horizontal localization performance and over time. Specifically, a cardioid microphone could decrease front/back errors over time, whereas left/right errors increased when different microphones were fitted to left and right ears. Front/back confusions were generally prominent. Objective measurements of interaural differences on KEMAR explained significant shifts in left/right errors. In conclusion, there is scope for improving the sense of localization in hearing aid users.

In the ear hearing device with a valve formed with an electroactive material having a changeable volume and method of operating the hearing device

For particularly good adaptation to a given hearing situation, an in-the-ear hearing device which has a housing with a channel in the housing that is designed as a through-opening for sound and air between the interior of the ear and the environment outside the ear, the channel is provided with a structural element for changing the size of the through-opening at at least one position. The structural element is a valve formed with electroactive material and the size of the through-opening is adjusted by application of a voltage to the valve.

Hearing aid having an operating device

The operation of a hearing aid (10) or hearing aid system (10, 11) is to be improved. It is proposed here that the acoustic auditory environment in which the hearing aid (10) or hearing aid system (10, 11) is found is analyzed and one of the adjustment functions dependent on the relevant auditory situation is assigned to at least one control element (7; 12-15) as a function of the auditory situation detected in this way. The adjustment possibility of the hearing aid (10) is thereby restricted to the adjustment possibilities which are meaningful to the relevant auditory situation.

A high performance pocket-size system for evaluations in acoustic signal processing

Custom-made hardware is attractive for sophisticated signal processing in wearable electroacoustic devices, but has a high initial cost overhead. Thus, signal processing algorithms should be tested thoroughly in real application environments by potential end users prior to the hardware implementation. In addition, the algorithms should be easily alterable during this test phase. A wearable system which meets these requirements has been developed and built. The system is based on the high performance signal processor Motorola DSP56309. This device also includes high quality stereo analog-to-digital-(ADC)- and digital-to-analog-(DAC)-converters with 20 bit word length each. The available dynamic range exceeds 88 dB. The input and output gains can be adjusted by digitally controlled potentiometers. The housing of the unit is small enough to carry it in a pocket (dimensions 150 × 80 × 25 mm). Software tools have been developed to ease the development of new algorithms. A set of configurable Assembler code modules implements all hardware dependent software routines and gives easy access to the peripherals and interfaces. A comfortable fitting interface allows easy control of the signal processing unit from a PC, even by assistant personnel. The device has proven to be a helpful means for development and field evaluations of advanced new hearing aid algorithms, within interdisciplinary research projects. Now it is offered to the scientific community.

A high performance wearable signal processor system for the evaluation of digital hearing aid algorithms

Today’s commercially available digital hearing aids contain specialized hardware to keep the power consumption low. These specialized processors are not able to run arbitrary signal processing strategies. However, new audiological algorithms have to be assessed in the everyday acoustical environment of the hearing impaired before developing specialized hardware. Therefore, flexible experimental hearing aids, which include general purpose signal processors, are needed for R&D purposes. Such a system has been developed, based on the high performance signal processor Motorola DSP56303. The wearable signal processor units also include high‐quality stereo analog‐to‐digital (ADC) and digital‐to‐analog (DAC) converters with 20‐bit word length each. The available dynamic range exceeds 90 dB. The input and output gain can be adjusted by digital potentiometers. The housing of the device is small enough to carry the experimental hearing aid in a pocket (dimensions 150×80×25 mm). Software tools have been developed to...

Reduction of time‐domain aliasing in adaptive overlap‐add algorithms

The adaptive overlap‐add algorithm (OLA) is an attractive means for acoustical signal processing (e.g., filtering, compression), since it is computationally effective and provides magnitude and phase information. It performs well when the input data blocks, the DFT, and the filter impulse response are of suitably chosen lengths. If the filter frequency response is altered on‐line by an adaptation rule formulated in the frequency‐domain, the resulting impulse response will, in general, be too long. This results in serious distortions, due to time‐domain aliasing during the inverse DFT. Limiting the filter length by a filter design procedure needs much computation time, which is inhibitive for many acoustical applications. Time‐domain aliasing distortions are particularly disturbing when musical signals are processed, due to their nonharmonic nature. An algorithm is proposed which convolves the filter frequency response with a short window sequence, solely in the frequency domain. Thereby, the aliasing comp...