Hearing Aid Signal Processing Research Articles

Recent developments in signal processing for digital hearing aids

pproaches in signal processing research on digital hearing aids fall into four areas, which cover signal acquisition, amplification, transmission, measurement, filtering, parameter estimation, separation, detection, enhancement, modeling, and classification. The first area uses advanced signal processing techniques to characterize and compensate for various hearing impairments such as loudness and frequency selectivity loss. The second area consists of effective target signal enhancement and noise reduction, which includes adaptive microphone array technologies, spectral

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.

Digital Signal Processing Hearing Aids, Personal FM Systems, and Interference: Is There a Problem?

To determine whether digital signal processing (DSP) hearing aids produce conducted radio frequency interference that can affect the use of personal FM systems, to quantify the nature of any such interference, and to discuss practical remedies. Sixteen DSP hearing aids were used. Measurements were made of the spectral characteristics of any conducted radio frequency interference produced by each aid with FM shoe and 40 cm direct audio input (DAI) lead when the DAI facility was enabled. Measurements were made with the aid, shoe, and lead inside an electrically screened chamber. The effect of DAI lead length was also investigated with one of the hearing aids. Finally, some subjective listening tests were carried out by using different FM systems coupled to a number of the aids. All but four of the DSP hearing aids tested produced readily measurable interference, with some much worse than others. Levels of interference were high enough with some hearing aids to be likely to significantly impede signal perception when the radio frequency of the interference coincided with the radio frequency of the FM system. This usually occurs intermittently as a result of the processor design of most DSP hearing aids. The listening tests suggested that when personal FM systems are in use with some DSP hearing aids, the interference would be audible, unpleasant, and detrimental to audio quality. DSP hearing aids without low electromagnetic interference processors should not be fitted to clients if personal FM systems are expected to be used. Manufacturers of DSP aids should be encouraged to use low electromagnetic interference processors in their DSP hearing aid design. Meanwhile, FM systems should be used with DSP hearing aids in such a way as to ensure high received radio signal levels, and FM receivers should be switched off when not in use.

Assessing FM transparency, FM/HA ratio with digital aids

The advantages of frequency modulation (FM) systems are well recognized.' In an ideal situation, an FM system that is connected to a hearing aid (HA) maintains the frequency-output characteristics of the hearing aid and enhances the signal-to-noise ratio (SNR) of the listening environment. That said, a clinician who works with FM and hearing aids must evaluate if the addition of the FM results in the optimal use of the FM-HA combination. Specifically, the clinician must verify that: (1) the electroacoustics of FM and HA meet specifications (i.e., sufficiently broad bandwidth, no evidence of distortion, etc); (2) the coupling of the FM does not change the frequency response of the hearing aid in the various FM receiver positions (such as HA, FM, FM+HA) for the same stimulus input (i.e., transparency); and (3) the ratio of the FM output to the hearing aid output, when used in the FM+HA mode, is favorable. This last verification goal is also known as the SNR advantage of the FM in the FM+HA mode. h is i mportant for clinicians working with these two devices to know how to determine the FM/HA ratio and, when necessary, to adjust the sensitivity of the devices to achieve the optimal ratio. The implicit expectation is that the FM+HA mode will increase wearer satisfaction. Clinicians have sought to standardize the verification of FM systems when used with hearing aids. However, with the advent of digital signal processing (DSP) hearing aids, the parameters and protocols previously used to verify FMs and HAs may be inadequate. Digital processing can interact with the type and duration of the stimulus used, resulting in equivocal results. Furthermore, the type of FM system used, as well as the hearing aid analyzer used to measure transparency and FM/HA ratio, may affect the outcome of the verification. Kuk and Keenan evaluated the impact of stimulus duration and stimulus type on the reliability and validity of verification. 2 Integrating their findings, we will present in this article a three-step protocol for measuring both the transparency and the FM/HA ratio in one measurement session. Specifically, we will describe the protocol, provide

Development of a Test Environment to Evaluate Performance of Modern Hearing Aid Features

This article describes a new test environment and materials with the potential to measure performance differences among different hearing aid signal processing methods and features. Normative data suggest a linearly predictable increase in difficulty on a speech-in-noise task as the masker changes from random noise to multiple-talker speech, and the number of talkers increases. Data collected with normal listeners revealed no differences across four test sites for the single loudspeaker (Noise-Front) results and some differences across test sites for the multiple loudspeaker results. Room dimension differences among audiometric test enclosures and diffusion (or lack thereof) of the maskers in the vertical and horizontal dimensions of the sound fields appear to account for performance differences for the multiloudspeaker arrays, confirming the need to limit maskers to aperiodic signals in rooms with controlled ceiling height or to establish norms for each test environment such that results obtained in different enclosures can be compared.

A novel signal-processing strategy for hearing-aid design: neurocompensation

A novel approach to hearing-aid signal processing is described, which attempts to re-establish a normal neural representation in the sensorineural impaired auditory system. Most hearing-aid fitting procedures are based on heuristics or some initial qualitative theory. These theories, such as loudness normalization, loudness equalization or maximal intelligibility can give vastly different results for a given individual, and each may provide variable results for different hearing impaired individuals with the same audiogram. Recent research in characterizing sensorineural hearing loss has delineated the importance of hair cell damage in understanding the bulk of sensorineural hearing impairments. A novel methodology based on restoring normal neural representation after the sensorineural impairment is presented here. This approach can be used for designing hearing-aid signal processing algorithms, as well as providing a general, automated means of predicting the relative intelligibility of a given speech sample in normal hearing and hearing impaired subjects.

Using digital hearing aids to visualize real-life effects of signal processing

Digital signal processing (DSP) hearing aids have made many advances since they were first successfully introduced in 1996. Today’s premium DSP hearing aids use far more sophisticated algorithms to process sounds than those of a few years ago. Some examples are modern algorithms for compression, feedback management, noise reduction, and control of directional microphones. The focus on finding more sophisticated ways to process sounds may be one reason that digital hearing aids accounted for 66% of all hearing aid sales in the United States in 2003.1 As DSP technology continues to advance, it is being extended to hearing aid applications not directly related to sound processing. Such applications are already evident in the use of DSP hearing aids for in situ threshold measurement2 and self-diagnostic testing.3 Another new application of DSP is for sound pressure level (SPL) measurement.

State of the art in perceptual design of hearing aids

Hearing aid capabilities have increased dramatically over the past six years, in large part due to the development of small, low-power digital signal processing chips suitable for hearing aid applications. As hearing aid signal processing capabilities increase, there will be new opportunities to apply perceptually based knowledge to technological development. Most hearing loss compensation techniques in today’s hearing aids are based on simple estimates of audibility and loudness. As our understanding of the psychoacoustical and physiological characteristics of sensorineural hearing loss improves, the result should be improved design of hearing aids and fitting methods. The state of the art in hearing aids will be reviewed, including form factors, user requirements, and technology that improves speech intelligibility, sound quality, and functionality. General areas of auditory perception that remain unaddressed by current hearing aid technology will be discussed.

Biological basis of hearing-aid design.

We show that we can accurately model the auditory-nerve discharge patterns in response to sounds as complex as speech and ask how we may exploit this knowledge to test new strategies for hearing-aid signal processing. We describe the auditory-nerve representations of vowels in normal and noise-damaged ears. The normal representations are predicted well by a cochlear signal processing model originally developed by Carney (Carney, L. H. J. Acoust. Soc. Am. 93:401-417, 1993). Basilar-membrane tuning is represented by a time-varying narrow-band filter. Outer hair cell control of tuning is exerted by a nonlinear feedback path. We show that the effects of noise-induced outer hair cell damage can be modeled by scaling the feedback signal appropriately and use the model to test one strategy for hearing-aid speech processing. We conclude by discussing some aspects of future trends in biomedical engineering approaches to problems of hearing impairment.

Feedback cancellation in hearing aids

Harry Levitt was an early advocate of digital signal processing for hearing aids. His interest included not only algorithms for compression and speech enhancement, but also more practical issues such as obtaining the desired hearing aid versus frequency response at the ear drum. Acoustic feedback is one problem that can limit the maximum gain possible in a hearing aid. Feedback cancellation, in which the acoustic feedback signal is estimated and subtracted from the microphone input, allows for greater hearing-aid signal amplification, and feedback cancellation was included in the work that Harry supported in his research group. In this presentation, the effects of feedback on the hearing-aid response will be reviewed. Digital adaptive filter techniques for feedback cancellation will then be presented, along with measurements indicating the limitations of feedback cancellation.

Localization and speech-identification ability of hearing-impaired listeners using phase-preserving amplification

Hearing-impaired listeners experience increased difficulties recognizing speech and localizing sounds in adverse environments. This study investigated the benefits of signal processing in binaural hearing aids designed to preserve cues that accompany spatial location. The ability of listeners to localize click-trains in noise was tested, along with their ability both to localize and to identify words in noise (a dual task). Listeners experienced two types of bilateral hearing aid fittings: (1) a custom fitting that provided appropriate gain while also matching the phase measured near the tympanic membranes without the hearing aids, and (2) a conventional fitting (using the same hearing aid device) that provided the same gain with noncustom, linear phase. Testing occurred for each fitting immediately and following 3-months listening experience using a within-listener, within-device, randomized, single blind, crossover design. A rating-scale questionnaire was administered to assess perceived speech- hearing and spatial abilities. In the dual task, the listeners exhibited superior localization ability for the phase-preserving fitting initially and after 3-months experience. This advantage did not occur for the click-train localization task. Listeners rated their spatial abilities higher with the phase-preserving fitting, although little improvement was observed or reported for speech hearing.

Whereʼs the proof?

Dispensers and their patients have quickly embraced digital signal processing hearing aids. But how much of their popularity is due to actual results and how much to “hype”? Two prominent audiologists offer differing but complementary perspectives.

Just Noticeable and Objectionable Group Delays in Digital Hearing Aids

Group delay in a digital signal processing (DSP) hearing aid may be perceived as an echo in the sound heard by a wearer listening to his or her own voice, due to a combination of unprocessed sound received at the ear through head and air pathways and delayed sound reaching the eardrum through the hearing aid. Depending on the amount, this delay may be audible or objectionable and can even result in auditory confusion. This study presents results from 18 subjects listening to their own voices through a DSP hearing aid with a variable group delay. The subjects varied the length of the delay and determined the amounts that were noticeable and objectionable as compared to the undelayed condition, while listening to their own amplified voices. Results indicated that a delay of 3 to 5 msec was noticeable to the listeners in 76 percent of the trials, and a delay of longer than 10 msec was objectionable 90 percent of the time.

Digital signal processing in hearing aids

For individuals with a hearing impairment, digital signal processing offers the promise of implementing previously unavailable methods of manipulating sound to compensate for hearing loss. However, hearing aids also present a complex technical challenge for the incorporation of DSP technology. Modern hearing aids are powered by zinc-air cells, since this chemistry produces one of the highest energy densities of all battery technologies and results in the most power for the smallest size. However, the use of these cells places severe limitations on the DSP technology that may be incorporated into a hearing aid. The available operating voltage is 1.25 V and, in order to provide an acceptable battery lifetime for the consumer, the available current drain is severely limited. An additional design challenge is that the entire DSP hearing aid must fit into the human ear. This presentation will focus on the complexity of achieving DSP technology in hearing aids within these electronic and mechanical limitations. General discussion will focus on circuitry necessary to achieve DSP in a hearing aid and on the various considerations and limitations that must be taken into account when making design decisions.

Using the Frye SPL Fitting Software to Fit Hearing Aids Incorporating Nonlinear Signal Processing.

There is no doubt that the hearing healthcare profession is at the dawn of an exciting new era in the advancement of hearing aid technology. Within the last few years there has been a major shift from dispensing linear hearing aid technology into a new era of so many unique hearing aid circuit choices. Today there are so many choices, from improved and more sophisticated linear circuits, i.e., class D amplifiers, to almost completely adjustable nonlinear analog and most recently fully digital circuits! Keeping current with today's hearing aid technology challenges even the most dedicated hearing healthcare professional. In fact, the idea of keeping current is what best summarizes the primary focus of this manuscript. At Frye Electronics, we are continually striving to find better, easier and more helpful methods of evaluating the electroacoustic performance of a hearing aid. An example of this commitment is the experimental “digital speech” software approach for evaluating the performance of the new digital signal processing hearing aids that we demonstrated at this year's American Academy of Audiology convention. We believe that the new Frye/SPL fitting test will help dispensers keep up with today's new technology by providing hearing healthcare professionals with an important next step that is both a practical and easy-to-use fitting approach for utilization with all types of hearing aids.

Non-linear signal processing in digital hearing aids.

Three different non-linear digital signal processing algorithms were developed; LinEar, DynEar and RangeEar. All three provided individual frequency shaping via a seven-band low-power filterbank and compression in two channels. RangeEar and DynEar used wide dynamic range syllabic compression in the low-frequency (LF) channel, while LinEar used compression limiting. In the high-frequency (HF) channel, RangeEar used a slow-acting automatic volume control, while DynEar and LinEar used compression limiting. Wearable digital signal processing-based experimental instruments were used to evaluate the fitting algorithms under real world conditions with experienced hearing aid users. Evaluation included laboratory testing of speech recognition in noise and questionnaires on sound quality ratings. Results did not indicate one general good-for-all algorithm, but different algorithms resulting in preference and performance depending on the hearing loss configuration. Preference for any of the new algorithms could be predicted based on auditory dynamic range measurements. It was hypothesized that the different preferences were affected by different susceptibility to masking of HF sounds by amplified LF sounds.

Technical and Audiological Factors in the Implementation and Use of Digital Signal Processing Hearing Aids

Fully digital hearing aids, to be worn behind or within the ear, are coming on to the market. The new possibilities and challenges of such aids are both technological and audiological in character. This paper describes some of the challenges, and one manufacturer's approach to them. The signal processing structure in one current digital hearing aid is presented, and justified in audiological terms, as a background to discussions of (1) the relative urgency of further advances in technology and audiological knowledge, and (ii) the challenges to the practice of hearing aid dispensing. It is concluded that further advances in rehabilitative audiology are more pressing than further major technological progress, and that the potential client benefits of the new generation of digital hearing aids will only he realized if the challenges they present to software design and dispensing practice are recognized and met.

Technical and audiological factors in the implementation and use of digital signal processing hearing aids.

Fully digital hearing aids, to be worn behind or within the ear, are coming on to the market. The new possibilities and challenges of such aids are both technological and audiological in character. This paper describes some of the challenges, and one manufacturer's approach to them. The signal processing structure in one current digital hearing aid is presented, and justified in audiological terms, as a background to discussions of (i) the relative urgency of further advances in technology and audiological knowledge, and (ii) the challenges to the practice of hearing aid dispensing. It is concluded that further advances in rehabilitative audiology are more pressing than further major technological progress, and that the potential client benefits of the new generation of digital hearing aids will only be realized if the challenges they present to software design and dispensing practice are recognized and met.

Using maximum length sequence coherence for broadband distortion measurements on hearing aids

Measurement procedures using broadband noise stimuli for testing hearing aids have recently been standardized in an attempt to ensure that test results more closely reflect the ‘‘real-world’’ performance of hearing aids. Although a number of researchers have employed coherence measures made with broadband noise stimuli to characterize the broadband distortion of hearing aids, broadband distortion measurement methods have not yet been standardized. In this paper, it is demonstrated through simulations and measurements on hearing aids that similar results are obtained when coherence is measured using maximum length sequence (MLS) and conventional unbiased statistical-based methods. The calculation of single- and dual-channel MLS coherence are also formalized. Signal-to-distortion ratio (SDR) measurements for two automatic signal processing hearing aids are presented.

Method for fitting binaural hearing aids

For a hearing aid wearer to perform binaural sound localization and to utilize directional hearing in noisy environments, it may be important to maintain at audible levels the binaural cues (i.e., interaural time and level differences) present without the hearing aid(s) in place. A method of achieving this hearing aid fitting goal for use with a prototype digital signal processing hearing aid has been developed. The method includes two major steps: hearing aid equalization (HAE) and hearing loss compensation (HLC). HAE is achieved with an FIR filter, which equalizes the amplitude and phase insertion effects of the hearing aids and maintains the binaural cues with the hearing aid(s) in place. The HAE filter coefficients are obtained from insitu probe tube measures of aided and unaided test signals using optimal filter calculations. HLC is also achieved with an FIR filter and associated gain. The HLC filter coefficients are obtained by using a weighted least-squares filter design technique. The target response for the HLC filter is determined from measures of electrical signal levels in the hearing aid circuit during threshold tests and during reference signal presentations in the sound field. The HAE and HLC filters are convolved to produce a single filter. A description of the fitting procedure will be provided, as well as examples of its use with hearing impaired individuals.