Digital Signal Processing Hearing Aids Research Articles

Hearing Aids in the Real World: Typical Automatic Behavior of Expansion, Directionality, and Noise Management

Automatic DSP (digital signal processing) features, widely available in hearing aids today, are useful because they alleviate the need for the hearing aid wearer to manually adjust the hearing aid as listening conditions change. Although the theoretical basis for the design of these features may be sound, little is known about their behavior in the real world. Data logging offers a glimpse into the life of the individual hearing aid wearer, but there are no published data to date that provide a frame of reference for the interpretation of this information. Further, data logging in hearing aids provides only aggregate summaries for individual features, ignoring complex interactions including the differences between the left and right sides of a bilateral pair. The purpose of this study was to determine the typical behavior of three automatic DSP hearing aid features-expansion, directionality, and noise management-in daily life. Ten individuals with hearing impairment were fitted bilaterally with BTE (behind the ear) hearing aids. The hearing aids were programmed for the individual's hearing loss with expansion, directionality, and noise management set to activate automatically. A PDA (personal digital assistant) logged the input level and status of expansion, directionality, and noise management from both devices at 5 sec intervals. Data were gathered in this manner over a period of 4-5 wk. A total of 741 hr of hearing aid use were logged, 50% of which were spent in environments no louder than 50 dB SPL. Expansion, directionality, and noise management were active 45, 10, and 21% of the time, respectively; the median amount of gain reduction for noise management was ∼1 dB. Although expansion and noise management were always active at the low and high input levels, respectively, activation of directionality never exceeded 50%. Expansion and noise management were sometimes active simultaneously, as were directionality and noise management. Bilateral agreement in feature activation typically exceeded 80%, except when the input level was at the cusp of a threshold for activation of a specific feature and at high input levels.

Multichannel Compression Hearing Aids: Perceptual Considerations

Multichannel Compression Hearing Aids: Perceptual Considerations

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.

Effects of Stimulus Level on the Speech Perception Abilities of Children Using Cochlear Implants or Digital Hearing Aids

The present investigation was designed to provide information to facilitate the decision of whether a child should continue using digital signal processing (DSP) hearing aids with wide dynamic range compression (WDRC) or be recommended for a cochlear implant, based on the unaided pure-tone average (PTA at 500, 1000, and 2000 Hz). Fifty-two children (ages 5 to 15 yr) with unaided PTAs in the moderately severe to profound range, wearing (DSP) hearing aids with (WDRC) or a Nucleus 24, Clarion 1.2, or CII cochlear implant system, participated: 26 with unaided PTAs from 60 to 98 dB HL using DSP hearing aids and 26 with pre-implant unaided PTAs from 93 to 120 dB HL, using cochlear implants. An open-set speech perception test, the Lexical Neighborhood Test (LNT; ), was administered at intensity levels representative of raised (70 dB SPL) and soft (50 dB SPL) speech at two different times approximately 1 mo apart. Minimum audibility of soft sounds was determined for the children with implants and with DSP hearing aids using warble-tone thresholds at octave intervals between 250 and 4000 Hz. Regression analyses and significance testing were used to determine the unaided PTA values at which the performance of the DSP Hearing Aid group (DSP HA group) and Cochlear Implant group on the LNT test were statistically different at the 0.05 significance level. For the 70 dB SPL presentation level, the statistically different PTAs were 113 and 97 dB HL at Time 1 and Time 2, respectively, and 96 and 88 dB HL at 50 dB SPL for Time 1 and Time 2, respectively. The Unaided PTA at which children in the cochlear implant group would be expected to score significantly better than the children in the DSP HA group was lowest (96 and 88 dB HL) for the lower signal level (50 dB SPL). Assuming that LNT scores at 50 dB SPL are representative of long-term hearing of soft incidental speech that is essential for language learning and fluent communication, the children with PTA values greater than the range from 88 to 96 dB HL would be expected to have significantly better LNT scores with a cochlear implant. These results should be further examined with research efforts focusing on early intervention with optimally fitted DSP hearing aids and cochlear implants.

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.

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.

The Hearing Aid Revolution: Fact or Fiction?

Objective – To present the experiences of consumers with traditional hearing aids (T-HAs) and digital signal processing hearing aids (DSP-HAs) during the period 1999–2001, based on data obtained from an ongoing quality assurance programme.Material and Methods – A questionnaire is mailed to subjects fitted with HAs 3–4 months after the fitting, and includes questions concerning satisfaction with the HA, frequency of use of the HA, ability to manipulate the HA, satisfaction with the overall services of the department and need (if any) for additional appointments.Results – The response rate was 69.5%, and thus information was obtained from 14 325 subjects (44.9% male, 55.1% female; median age 72 years; range 18–97 years). Of the respondents, 7983 (55.7%) had been provided with T-HAs and 6342 (44.3%) with DSP-HAs. The results of the questionnaire were as follows: 71.4% of those fitted with a T-HA were very satisfied/satisfied with it, compared to 68.1% with a DSP-HA (p<0.05); 91.6% and 89.1%, respectively used their HA daily/weekly (p<0.05); 80.5% and 82.2%, respectively, were able to manipulate their HA; 96.2% and 97.3%, respectively were satisfied with the overall services of the department; and 32.5% and 48.5%, respectively, indicated a need for an additional appointment (p<0.05). A comparison between high- and low-cost DSP-HAs showed that 68.4% and 68.2%, respectively were very satisfied/satisfied with their HA (p=NS).Conclusions – According to consumers the “HA revolution” has failed to materialize; the significantly higher proportion of subjects with DSP-HAs who need an additional appointment represents a heavy burden on the hearing health services. The lack of a difference between the benefits obtained with low- and high-cost DSP-HAs emphasizes the need for appropriately designed and performed trials before new HA technology is launched.

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.

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.

DSP algorithms for hearing aids

Advances in semiconductor technology are making digital signal-processing (DSP) hearing aids a viable alternative to conventional analog devices. Some of the most promising DSP algorithms for hearing aids are found in the areas of (1) microphone-array processing for interference reduction, (2) feedback reduction for increased gain margin and stability, and (3) automatic gain control for improved audibility and comfort. Typical common elements of such algorithms are a tapped-delay line, which is used to store a sequence of input values sampled over time, and a time-varying response, which is computed based on the input values. For example, adaptive filters are required by feedback reduction techniques that estimate the feedback path, as well as by adaptive microphone arrays, which attenuate spatially distinct interference sources. Additional benefits in DSP hearing aids are expected from their ability to use sophisticated fitting methods, particularly when the fitting procedure includes selection of algorithm-specific parameter values as well as the frequency gain characteristic. Finally, DSP hardware platforms designed for the hearing aid application provide a powerful research tool, facilitating field trials designed to evaluate algorithms and determine optimal parameter settings. In particular, such devices allow side-by-side comparison of multiple algorithms, including both DSP algorithms and those that could ultimately be implemented using analog components. [Work supported by NIH.]

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.