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Resound - Hearing Experience - Feb. 2021

Behind The Hype: Digital Amplification

Behind The Hype: Digital Amplification
Giselle Matsui
January 28, 2002
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This article is sponsored by ReSound.
Overview:

We are all besieged with marketing claims of products that allege to make our laundry whiter, our dusting easier and our thighs thinner. Many times, because we are not experts in those fields, we simply evaluate products based on their marketing claims and make our best guess regarding whether to purchase, or not. Sometimes we are pleasantly surprised and sometimes we realize we have wasted our money.

In our industry, new hearing instruments and technologies seem to arrive daily, coupled with amazing marketing claims. Most professionals in the hearing instrument industry have years of education and experience to fall back on when evaluating impressive claims from new technology. As licensed healthcare professionals, our patients trust their hearing healthcare to us, and therefore it is our responsibility to look behind the marketing claims.

The goal of this article is to provide a framework from which to evaluate new hearing instrument technologies. Remember that when trying new products on a patient, professionals need to demonstrate confidence and knowledge specific to that product, which will dictate the choice of technology. To do this, professionals must have the ability to discern between technology features and glossy brochures.

DIGITAL vs. ANALOG

Since the introduction of hearing aids, professionals have struggled to determine which technology is truly best for their patients. The digital products available in 2002, have tremendously increased patient's options and alternatives, particularly when compared to what was available 4 or 5 or 6 years ago. Early commercially available digital solutions appeared to some, to have been no better than the analog based programmable instruments available at that time.

However, the digital "buzzword" resulted in patients asking for digital technology, and helped the professionals and the manufacturers realize that the future was digitally based. Once manufacturers were able to minimally harness the power of digital chips, digital hearing instruments provided technology options that analog instruments simply could not. Of course the harnessing of digital ability is nowhere near complete, even in 2002! Research and development will continue and very probably the digital technologies which will be available in 2007, cannot even be imagined today!

Nonetheless, there are clear-cut differences between analog and digital instruments.

Analog:

The strength of analog based instruments is that they are able to provide excellent WDRC compression systems coupled with 2 or 3 band frequency resolution shaping. Generally speaking, analog based instruments are less expensive than digitally based systems, but that too, is changing rapidly. Digital systems are becoming less expensive as the manufacturers realize the advantages of "economies of scale." Analog based systems are typically not as refined as digital systems. Nonetheless, analog systems work well for patients with non-demanding listening needs, and non-demanding life styles.

Digital:

Digital hearing instruments provide a wide range of highly sophisticated signal processing systems which can enhance speech recognition in noise and create a more comfortable listening environment (Agnew, 1999, Edwards, 2000). Most digital instruments manipulate the input signal in ways not possible with analog instruments (Edwards et al, 1998, Olson et al, 2001, Powers, 1999),

Patients who have more demanding listening needs should be fit with more advanced digital signal processing systems. Importantly, one of the primary advantages of digitally based hearing instruments is their ability to perform many sophisticated functions simultaneously. Simply, analog technologies cannot accomplish this.

Noise Reduction Issues:

One of the major complaints consistently reported by patients wearing hearing aids, is that hearing aids reproduce too much background noise. It is important to note that hearing aids don't create background noise. It is more likely that the hearing impaired person wearing hearing aids, is able to hear the background noise that they have long since stopped hearing! Further, because they are not accustomed to hearing background noise they cannot "tune it out" readily, nor does their efferent auditory nervous system "squelch" background noise as would be expected for a person with normal hearing.

Therefore, hearing impaired listeners, while wearing hearing aids, desire noise reduction options. These options are available in various forms and configurations with digitally based hearing aids. Additionally, more than one noise reduction technique can be applied, if and when desired, using digital technologies.

Noise Reduction: Historical Notes

The introduction of analog based ASP (automatic signal processing) instruments in the mid-1980's ushered in the trend of manufacturers attempting to control the level of background noise, essentially through low frequency gain reduction. This gain reduction occurred after a compression knee-point had been exceeded and many times resulted in a loss of speech intelligibility. Even after purchasing ASP technology, patients came to our offices for relief from background noise, or relief from the resulting decrease of gain!

Analog programmable instruments gave us a little more ammunition against background noise. Typically, we could create a second environmental program in a multi-memory instrument with reduced low frequency gain and enhanced high frequencies. The patient was instructed to use this memory in noisy situations. This was more effective than conventional ASP instruments. However, we were still restricted to frequency and gain manipulations and patients had to manually change the instrument into the noise management memory.

Noise Reduction: Digital Noise Management

Digital technology has truly opened up the world of background noise management in hearing instruments. Identification of speech vs. noise, multiple bands and multiple time constants all contribute to effective noise management (Agnew, 1999). Digital noise reduction systems will greatly enhance patient comfort while wearing a hearing instrument in background noise and reduce overall listener fatigue.

Before any noise reduction algorithm can be effective, the hearing instrument must first determine whether there is speech or noise in any given band.

There are currently two methods for determining if a signal is speech or noise: monitoring modulation differences or monitoring changes in harmonic speech energy. Both appear to work equally well. As soon as the instrument determines the incoming signal is noise, the algorithm applies a gain reduction strategy in the band which contains the noise element. This is where differences in instruments appear.

Noise Reduction Characteristics:

The two most important characteristics in noise reduction strategies are the number of bands (and the subsequent bandwidths) and the time constants associated with the gain reduction.

Frequency Bands and Noise Reduction:

If an instrument has only a few frequency bands, then each band represents a relatively larger portion of the spectral energy and any gain reduction impacts a large portion of the frequency response. This typically does not permit the noise reduction to be active at the same time speech is present or worse, only allows the noise reduction to be active and thereby reduces the gain for all signals -- including speech.

With more frequency bands, the noise reduction can occur over smaller segments of the frequency response. This allows speech to pass unchanged in some parts of the frequency response while undesirable noise is reduced in another part of the frequency response.

Time Constants and Noise Reduction:

The second important characteristic of a noise reduction algorithm are the time constants associated with the onset and release of the noise reduction system. In this arena, faster time constants produce a more pleasing type of noise reduction. Slow time constants may cause the patient to lose elements of the speech signal during the onset or release of noise reduction. Fast time constants allow the noise signal to be reduced between words and many times between syllables. Perceptually the patient will notice a drop in the background noise while the speech signal remains at a comfortable level. This provides a very comfortable listening experience for the patient, even in a noisy environment.

Directional Systems - Variations on a Theme:

Not all directional microphone systems are the same. There are basically 2 types of directional microphone systems: one is based on single microphone with two acoustic ports and the other on two separated omni microphones with one port each. Historically, single microphone directional systems utilized one microphone with introduced time delay between two signals coming from two ports, which were physically separated from each other. Accordingly, the port that was farther from the sound source was going to receive the signal later than the nearer port. The amount of improvement in the signal to noise ratio for signals coming from the front (0° incident) over the signals that arrive equally from all other directions is known as the directivity index (DI) (Wolf, Hohn, Martin, Powers, 1999). This type of system offered a fairly low DI of around 3-4 dB (Edwards, 2000). This type of system did not offer the patient the ability to choose between omni directional and directional.

This problem was solved with combination called D-MIC™, which utilized one omni and one directional two ports microphone. Another part of the system was some sort of equalizer, which was used for LF boost in directional mode. Having omni and directional microphone, allowed switching between two modes, depending on the patient's needs.

Devices that utilize dual microphone directional systems also allow the patient to choose between omni-directional and directional modes. There are two types of dual microphone systems. The first type of system incorporates two omni directional microphones and routes the signal from the two microphones into one pathway and utilizes the physical distance between the two microphones and/or an electrical delay to create a directional polar pattern. The attenuation associated with the polar pattern will reduce background noise coming from the rear plane.

The above dual microphone system, along with the D-MIC, creates a single fixed polar pattern as dictated by the manufacturer. Typically, if the manufacturer utilizes the cardioid pattern, the device will have a DI of approximately 3 - 4.8 dB. Likewise, if a hypercardioid pattern is utilized the DI will be up to 6 dB when the noise source is located in the null of the polar pattern. As the noise source moves away from the null, the amount of attenuation is reduced. These two types of directional system are available in both analog and digital processors.

The second type of digital dual microphone system, also utilizes two omni-directional microphones. However, the second type routes the signal from each microphone through two separate A/D converters. With digitally programmable delay, the device can be programmed for different polar patterns depending on the patient's needs. The most advanced manipulation of these signals resulted in a feature that is known as Adaptive Directionality. Adaptive Directionality allows a hearing instrument to have an infinite number of polar patterns, because the system places the null of the directional pattern towards the loudest signal in the rear plane of the patient. This system is the most reactive to the patient's environment and creates the optimal directivity index regardless of the location of the noise source since the noise source is always located in the null. This type of system is only available in digital devices.

When choosing a directional system, it is important to look at your patient's lifestyle. If your patient reports a fairly sedentary lifestyle, then a good analog or moderate DSP WDRC instrument with an omni directional or a single directional microphone may serve their needs very well. Patients who find themselves in more demanding listening environments such as meetings, bowling alleys or Bingo halls will most likely receive more benefit from an adaptive directionality DSP dual microphone system. I recommended that any previous hearing aid user who has had background noise complaints should be fit with adaptive directionality.

FEEDBACK SUPRESSION:

Feedback suppression is another area in which digital hearing instruments truly have a distinct advantage over analog hearing instruments. Feedback suppression in analog instruments requires gain reduction. Typically, when feedback is present in an analog-based system, the professional is instructed to reduce the gain in the high frequency band(s). In a two or three-channel instrument, this can seriously impact the high frequency response of the hearing instrument and the patients perception of high frequency information.

Many digital hearing instruments also utilize gain reduction based strategies. Fortunately, digital instruments may offer a greater number of bands or finer resolution notch filters, resulting in less impact on the frequency response. However, there is still an adverse affect on the high frequency response of these instruments due to the gain reduction.

True digital feedback suppression involves the production of a counter-phase signal to nullify any feedback signal before it becomes audible. Typically, these instruments make some measurement of the feedback path with the instrument in situ so that it is customized for each patient. Based upon that measurement, the instrument creates a signal which is opposite in phase to the feedback signal and adds it to the input. In this way, the feedback signal is simply cancelled. These algorithms typically give an 8 -12 dB increase in available gain that the patient would not have had with traditional methods of feedback reduction. These instruments may also have an adaptive element to the algorithm so that they will not feedback in a changing environment, such as holding a telephone to the aid. This is the most effective method of feedback reduction since there is no affect on the frequency response and it adapts to various environments.

Some feedback management systems also are able to take the information from the measurement and visually represent in the software the amount of gain available before feedback. This powerful tool can help identify when and if feedback is going to be a problem and guide the dispenser in making reliable adjustments to the frequency response or vent choices.

SUMMARY:

In the ever-changing world of hearing healthcare and related technology, the hearing healthcare professional must determine which technology is most appropriate for each individual patient. It is necessary to have a clear understanding of the advantages and disadvantages of all the instruments we choose to fit.

Creating a questions-based approach to help determine the technology most appropriate for each individual patient is an important step in any hearing aid fitting. When choosing a digital amplification system, you may want to consider the following questions concerning your patient's needs:

1. Is my patient bothered by background noise and might this patient require a noise reduction system?
  • Does the system differentiate between speech and noise?

  • How many bands does the instrument have?

  • What are the time constants of the noise reduction?

  • What are my patient's comfort issues?


  • 2. Does my patient have difficulty understanding speech in back ground noise?
  • Does the system have a single microphone system or a dual microphone system?

  • What is the typical Directivity Index of the system?

  • Does the system have a single fixed directional pattern or does the directional pattern adapt to the environment?

  • Can my patient choose omni directional or directional depending on the environment?


  • 3. Does my patient have a history of feedback problems due to physical characteristics of the ear, audiometric configuration, or gain requirements?
  • Does this instrument use a gain reduction approach to feedback management? If so, will it have a detrimental affect on my patient's speech intelligibility due to high frequency gain reduction?

  • Does the feedback reduction system measure the patient's feedback path in situ for a customized solution?

  • Is there an adaptive element to the feedback reduction system?


  • REFERENCES:

    Agnew, J: Challenges and Some Solutions for Understanding Speech in Noise. High Performance Hearing Solutions, 1999, Vol. 3: 4-9.

    Edwards B :Beyond Amplification: Signal processing techniques for improving speech intelligibility in noise with hearing aids. Seminars in Hearing, 2000; 21 (2): 137-156.

    Edwards B, Struck C, Dharan P and Hou Z: New digital processor for hearing loss compensation based on the auditory system. Hear J 1998;51 (8): 38-49.

    ER-81A D-MIC™ Capsule Application Note #1. Etymotic Research, 61 Martin Lane, Elk Grove Village, Illinois 60007

    Olson L, Musch H and Struck C: Digital Solutions for feedback control. Hearing Review 2001; 8 (5): 44-49.

    Powers, T and Wesselkamp, W. The Use of Digital Features to Combat Background Noise. High Performance of Hearing Solutions 1999, Vol. 3: 36-39.

    Wolf, R, Hohn, W, Martin, R, and Powers, T. Directional Microphone Hearing Instruments: How and Why They Work. High Performance of Hearing Solutions 1999, Vol. 3: 14-25.
    NEW 4-part series presented in partnership with the National Acoustics Lab | October 5, 12, 19, + 26 | 5:00 pm EDT | Course Typ

    Giselle Matsui

    Director of Product Management, GN ReSound



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