Digital Signal Processing Hearing Research Articles

Audibility and Speech Perception of Children Using Wide Dynamic Range Compression Hearing Aids

This study examined the relation of audibility for frequency-specific sounds and the Speech Intelligibility Index (SII) to speech perception abilities of children with sensorineural hearing loss using digital signal-processing hearing aids with wide dynamic range compression. Twenty-six children age 5-15 years with pure-tone averages (0.5, 1.0, and 2.0 kHz) from 60-98 dB HL participated. Three subgroups were created based on the compression characteristics of each hearing aid. Minimum audibility was determined using aided thresholds for frequency-modulated tones and the SII calculated at 55 and 70 dB SPL using the simulated real-ear output of the hearing aid. The Lexical Neighborhood Test (LNT; K. I. Kirk, D. B. Pisoni, & M. J. Osberger, 1995) was presented at 50 and 70 dB SPL. LNT scores at 70 dB SPL were significantly higher than at 50 dB SPL. Average aided thresholds at 0.5, 1.0, and 2.0 kHz were negatively correlated with LNT scores at 50 dB SPL, and SIIs at 55 and 70 dB SPL were positively correlated with LNT scores at 50 and 70 dB SPL. Results support using aided thresholds and speech test scores at soft to loud levels as part of the amplification fitting process.

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.

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.

Artificial noise signals with speechlike spectral and temporal properties for hearing instrument assessment

The introduction of different forms of nonlinear amplification and noise-reduction algorithms calls for more intelligent ways to specify the properties of hearing instruments. Traditional specifications no longer reflect important characteristics of hearing instruments. Also, for the psychophysical evaluation in a laboratory setting, well-defined noises are necessary; aided speech perception will be largely determined by the properties of the speech signal and the concurrent background noise. The field of hearing instrument specification and evaluation has become a complex area in which traditional tools have lost their relevance. There is a strong need for a new approach based on new test signals that are broadband and dynamic. Standardized broadband noises are required for comparative measurements in different hearing instruments. Based on these notions a working group of the International Collegium of Rehabilitative Audiology (ICRA) started a consensus process to design well-defined artificial noise signals that can become a standard for hearing instrument assessment, both for technical measurements and for applications in the clinical and psychophysical evaluation. This paper reports about the design criteria, the actual realization, and some examples of applications in the measurement of digital signal processing hearing instruments.

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.

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.

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.

Feedback suppression in digital signal processing hearing aids

Acoustic feedback in digital signal processing hearing aids is suppressed by using signal processing techniques in the digital processor. A first processing technique causes the data to the main signal processing path in the digital signal processor to be delayed by varying amounts over time, preferably in a periodic manner, to disrupt the buildup of feedback resonances. In a second technique, a digital filter receives the input data and has its coefficients adjusted so that the output of the filter is substantially an optimal estimate of the current input sample based on past input samples. The output of the filter is then subtracted from the input signal data to provide difference signal data which substantially cancels out the resonant frequencies. In a third technique, the acoustic feedback path from the output to the input of the hearing aid is modeled in the digital signal processor as a delay and a linear filter. The output of the main signal processing path in the digital signal processor is delayed and the delayed data passed through the linear filter, with the output of the filter then being substracted from the input signal data to provide difference signal data which is provided to the main signal processing path. The coefficients of the digital filter in the feedback path are adjusted so that the signal passed through the feedback filter substantially corresponds to of the acoustic feedback signal to thereby cancel the same.

Evaluation of two voice-separation algorithms using normal-hearing and hearing-impaired listeners.

Two signal-processing algorithms, designed to separate the voiced speech of two talkers speaking simultaneously at similar intensities in a single channel, were compared and evaluated. Both algorithms exploit the harmonic structure of voiced speech and require a difference in fundamental frequency (F0) between the voices to operate successfully. One attenuates the interfering voice by filtering the cepstrum of the combined signal. The other uses the method of harmonic selection [T. W. Parsons, J. Acoust. Soc. Am. 60, 911-918 (1976)] to resynthesize the target voice from fragmentary spectral information. Two perceptual evaluations were carried out. One involved the separation of pairs of vowels synthesized on static F0's; the other involved the recovery of consonant-vowel (CV) words masked by a synthesized vowel. Normal-hearing listeners and four listeners with moderate-to-severe, bilateral, symmetrical, sensorineural hearing impairments were tested. All listeners showed increased accuracy of identification when the target voice was enhanced by processing. The vowel-identification data show that intelligibility enhancement is possible over a range of F0 separations between the target and interfering voice. The recovery of CV words demonstrates that the processing is valid not only for spectrally static vowels but also for less intense time-varying voiced consonants. The results for the impaired listeners suggest that the algorithms may be applicable as components of a noise-reduction system in future digital signal-processing hearing aids. The vowel-separation test, and subjective listening, suggest that harmonic selection, which is the more computationally expensive method, produces the more effective voice separation.