Showing papers on "Digital hearing aid published in 1990"
01 Nov 1990
TL;DR: A method for adaptively equalizing the ubiquitous feedback path of a hearing aid in order to stabilize the system and an additional 10 to 15 dB of stable gain margin has been demonstrated.
Abstract: A method is described for adaptively equalizing the ubiquitous feedback path of a hearing aid in order to stabilize the system. The algorithm utilizes an LMS adaptive filter and is implemented in digital form. An additional 10 to 15 dB of stable gain margin has been demonstrated.
25 citations
TL;DR: The use of the digital hearing aid as a simulator of other hearing aids is discussed and an illustrative example provided in which a new form of amplitude compression, orthogonal-polynomial compression, has been simulated using a digital master hearing aid.
Abstract: Digital hearing aids offer many advantages over conventional hearing aids. These include signal-processing capabilities that are superior to those of a conventional analog hearing aid, methods of signal-processing and control that are unique to digital systems and which cannot be implemented in conventional analog hearing aids, and innovative new techniques that are changing our way of thinking about hearing aids. An example of the first of these advantages is the extremely high precision with which the frequency-gain characteristic can be specified and the use of this capability to study the effects of frequency response irregularities commonly encountered with hearing aids. An example of the second advantage is the use of memory and logical operations in the implementation of multivariate adaptive paired-comparison techniques for more effective hearing-aid prescription. Another example is the use of powerful new signal processing techniques for noise reduction. The third advantage is the most important. The digital hearing aid can be viewed as a generalized hearing instrument which can be used for simulation, testing and prescription, as well as amplification. The use of the digital hearing aid as a simulator of other hearing aids is discussed and an illustrative example provided in which a new form of amplitude compression, orthogonal-polynomial compression, has been simulated using a digital master hearing aid.
19 citations
TL;DR: The results indicate that loudness compensation without highpass filtering improves speech intelligibility for a wide range of input levels by achieving an optimum gain characteristic at all input levels.
Abstract: Several attempts that employed dynamic compression to compensate for cochlear impairment in combination with recruitment have failed to improve speech intelligibility. One reason is the attenuation of dynamic intensity modulations in the range of 1 to 50 Hz which contribute significantly to speech discrimination. In the approach presented here, these modulations were enhanced rather than attenuated by an envelope highpass filter operating in the loudness domain as follows: the output of a filterbank was lowpass filtered to extract the envelope, converted to loudness in accordance with normal hearing, highpass-filtered and converted back to the 'desired' envelope by the loudness characteristic of each individual hearing-impaired subject. The output signal was finally obtained by summing the outputs of the filterbank channels after correcting their amplitudes. The algorithm was implemented by a computer to simulate a digital hearing aid fitted to the loudness-scaling function of each individual. Loudness scaling and speech-discrimination data in quiet and in noisy conditions are presented for 2 impaired subjects with and without highpass filtering in the loudness domain. The results indicate that loudness compensation without highpass filtering improves speech intelligibility for a wide range of input levels by achieving an optimum gain characteristic at all input levels. The highpass filter applied to the envelope has no noticeable effect on speech discrimination either in quiet or in noise.
7 citations
TL;DR: The place of speech intelligibility considerations in the ensemble of criteria that could be associated with these perceptual attributes of speech naturalness, intelligibility, pleasantness, etc is discussed.
Abstract: With the use of sophisticated speech processing techniques in auditory prosthetic devices, it will be possible to convert the speech signal to maximal satisfaction for the user. However, the user's satisfaction is a multidimensional percept that may include speech naturalness, intelligibility, pleasantness, etc. This paper discusses the place of speech intelligibility considerations in the ensemble of criteria that could be associated with these perceptual attributes. It also discusses some of the problems that may be responsible for the relative inadequacy of the available speech intelligibility prediction procedures for use in digital hearing aids.
2 citations
TL;DR: A wearable speech processing hearing aid was programmed as a linear amplifier for two subjects with rapidly sloping profound high‐frequency hearing loss and the maximum power output was programmed to be below the uncomfortable listening level.
Abstract: Acoustic tones were delivered using a hearing aid receiver in the custom earmold of the subject. A probe‐tube microphone was used to measure the sound pressure level (SPL) in the ear canal with the earmold and receiver in place. In‐the‐ear SPLs corresponding to threshold, soft, comfortable, loud, and uncomfortable listening levels were recorded. Using these data, a wearable speech processing hearing aid (WSPHA) was programmed as a linear amplifier for two subjects with rapidly sloping profound high‐frequency hearing loss. The WSPHA was implemented using a 24‐bit digital signal processor (DSP56001) with 14‐bit analog‐to‐digital and digital‐to‐analog converters. The WSPHA is adjustable to within 1 dB of any specified frequency response. The fitting of the WSPHA was based on matching the long‐term speech spectrum to the contour of the comfortable listening level. The maximum power output was programmed to be below the uncomfortable listening level. Preliminary speech perception results indicated that the sub...
1 citations
TL;DR: In this paper, a method of designing a digital hearing aid for speech signal suitable for sensorineural hearing loss with a narrow audible range is proposed, where the input signal is divided into short time blocks (8 ms) and its spectrum is analyzed using short-time FFT.
Abstract: A method of designing a digital hearing aid for speech signal suitable for sensorineural hearing loss with a narrow audible range is proposed. First, the input signal is divided into short time blocks (8 ms) and its spectrum is analyzed using short‐time FFT. Then the frequency‐gain characteristic of the digital filter for the block is determined using loudness compensation functions that describe the relation between the loudness of normal listeners and that of impaired listeners, so that the input signal within the block is projected on to the dynamic range of the subject. This algorithm is realized using a digital signal processor (DSP). The results of monosyllabic intelligibility tests to evaluate the performance of the system for six sensorineural impaired listeners are also shown. [Work supported by Grant‐in‐Aid for Scientific Research from the Ministry of Education and the Ministry of Public Welfare.]
1 citations