New Widex Fitting Tutorial Video, Using Compass Step By Step To Perform A Fitting.

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This new video is one of a series that has been released in the last few days. We shall feature each one.

Guide To Using The Audibility Extender Successfully

This is a guide explaining the use of the Widex cornerstone audiological strategy of Frequency Transposition. This feature is named The Audibility Extender

You may download the file here Guide To Using The Audibility

Widex M Advert,

I still think this advert drips style and sophistication.

Talking about Tinnitus

New Presentation, Lets Talk About Tinnitus

[slideshare id=6869226&doc=lets-talk-tinnitus-1297296259-phpapp02]

Talking About Evidence, Zen & Tinnitus

Sweetow Commercial, Zen & Tinnitus

WidexLink

WidexLink
- a New, Industry-leading Technology for Wireless Transmission

In recent years, we have seen a variety of new hearing aids on the market with different wireless communication abilities. Some offer the possibility of transmitting sound to the hearing aid from a mobile phone, a TV, or an MP3 player. Others hearing aids are capable of exchanging coordination data, and a small number are able to transmit sound to one another.

We believe that the key to new achievements in the hearing aid industry lies in the ability to use a sophisticated wireless transmission technology which will facilitate the coordination of dynamic settings between two hearing aids, high quality transmission of audio from external sound sources, and wireless transmission of data between two hearing aids.

We are convinced that wireless technology is the future in the hearing aid industry. However, the success of any given wireless transmission technology will depend on at least three key issues when applied to hearing aids: Sound quality, transmission robustness, and power consumption.

WidexLink is our new proprietary digital radio frequency transmission technology. The new technology has been designed to provide the highest audio quality and efficiency. WidexLink offers new possibilities for extended bandwidth audio transmission between hearing aids (for example, CROS and BiCROS applications),extended bandwidth audio streaming from external assistive listening devices (for example, DEX devices) to hearing aids, and the continuous exchange of synchronisation data between hearing aids and external devices (Inter-ear communication).

The unique, digital wireless link offers an unparalleled low latency (delay) of <10 ms when transmitting audio. This ensures minimum distortion and echo-free audio quality when direct acoustic sound in the room is mixed with transmitted sound.

The need for a new technology

Today, Bluetooth is probably the most popular wireless technology for transferring data and digital sound between devices. The Bluetooth technology is available in a large variety of "plug-and-play" chip solutions. From an engineering point of view, therefore, Bluetooth is the fastest way to a digital wireless design. However, Bluetooth has some drawbacks which create serious problems in hearing aid applications.


First and foremost, Bluetooth is an extremely energy-demanding technology. The Bluetooth chip uses so much power that it has no practical application in the hearing aid industry. Moreover, even if another chip is used, battery drainage is still very high during Bluetooth transmission. Hearing aid manufacturers are therefore forced to find ways to optimise Bluetooth transmission if the power consumption is to be kept at a reasonable level.


Another serious drawback is that standard Bluetooth has a high built-in latency (delay) of 150 ms when transferring audio through the Bluetooth codec. That is, transmitted sound will reach the ear 150 ms later than the direct acoustic sound in the room. In hearing aid applications, this is a serious issue because the time delay between sound sources comprises important psychoacoustic cues about direction and distance to sound sources. The latency can be reduced by switching off some of the technological features while transferring digital audio, but there is a limit to how far this optimisation can be forced. As far as is known, the Bluetooth latency can be reduced no further than approximately 45 ms which is not sufficient to avoid artefacts in hearing aid applications.

Latencies and artefacts

A small delay of 1 to 10 milliseconds is unproblematic. However, when the delay between transmitted and direct sound becomes longer than 10 milliseconds, artefacts begin to occur. The first artefact to occur will be an audible comb-filter effect. It is called "comb-filter effect" because it filters away frequencies like a comb, making notches in the spectrum. The resulting sound will be hollow and metallic. Above 40 milliseconds the streamed sound will be perceived as an echo of the direct sounds. An additional problem can occur when sound is transmitted from for example a TV. When the delay reaches around 150 milliseconds, which is the default delay with Bluetooth, lip movements will begin to appear unsynchronised with the sound. Thus, hearing aid manufacturers who rely on Bluetooth or another technology which introduces a delay above 10 ms will not be able to avoid artefacts in their applications.

Also included on the list of Bluetooth drawbacks are the relatively large chip size and the high current consumption which are obviously essential parameters in a hearing aid design.

The artefacts that can occur as a result of the delay between direct sound and digital audio stream are listed in the figure below:

Figure 1. Psycho-acoustic artefacts due to latency (delay) between direct sound and transmitted sound.

 

The new technology from Widex has un-paralleled low latency of less than 10 ms between the direct acoustic sound and the digital audio stream. And in CROS and BiCROS applications, the delay is even shorter.

Advantages of WidexLink

It is no simple matter to design a system which can simultaneously maintain high audio quality, low battery drain, and robustness against transmission errors.
Widex achieved these goals by developing a very efficient audio coding method which is custom-designed for use in a hearing aid platform.


It was considered of the utmost importance that the highest possible audio quality should be maintained with the new wireless technology. In order to achieve this goal, several key components of the wireless system were carefully designed. First, a very efficient and robust audio codec (short for encoding/decoding) and radio frequency (RF) transmission system were developed to ensure fast, stable, and trouble-free transmission of sound and data during normal use.  It was a requirement that the digital audio codec be based on a coding principle which will ensure that the signal is perceived to be as close to the original sound as possible.  Another factor in maintaining a high audio quality was a low audio delay over the wireless transmission system.  To achieve this, the digital coding system was designed in such a way that no data would need to be re-transmitted and consequently slow down the transfer of the audio signal as a result.


Ensuring battery efficiency was also a major concern in the design of WidexLink. This goal was achieved through the employment of a very efficient data compression method, and the invention of a new, highly sensitive radio receiver which permits low transmission power.


Robustness was achieved in multiple ways. In addition to the inherent robustness of the audio codec and the radio receiver mentioned above, robustness was also attained by means of a highly efficient channel coding which detects and handles errors quickly and securely.


How WidexLink addresses the key factors will be discussed in more detail below.

High sound quality

Audio bandwidth

Audio bandwidth is one of the key factors in maintaining a high sound quality. Thanks to the new technology in the Clear product range, we are able to offer an exceptionally broad audio bandwidth in models with a Clearband receiver, stretching from 100 Hz to 11.2 kHz for transmitted sound. This is industry leading.

One area where a broad audio bandwidth makes a clear difference for hearing aid users is when they listen to music. The high frequencies provide ambience and brilliance to the sound. Thus, the sound experience will be somewhat richer with an upper bandwidth of 11 kHz when listening to the crispy sound of a hi-hat or cymbals, for instance. Similarly, an audio bandwidth stretching as far down as 100 Hz will produce a fuller bass.

Codec

Another way to maintain a high sound quality is to develop an efficient codec. Digital audio data as we know them from CDs are extremely bulky. It is therefore necessary to reduce digital audio data in some way as the transmission bandwidth is too narrow to effectively transmit the raw audio signal. This is achieved by means of data compression (not to be confused with the dynamic compression of the audio signal in the hearing aid).

Digital audio data compression is achieved by means of a set of complex algorithms called an audio codec.  An audio codec consists of two parts:  encoding and decoding.  The purpose of the audio encoding is to reduce the size of the digital data representing the original signal. The purpose of the decoding is to reconstruct the encoded audio signal in a manner which ensures that it is as close as possible to the original audio signal. This process is analogous to the shipping of a parcel by mail. If you wished to send an office desk from the U.S. to Europe, you could simply place the assembled desk inside a large box and send it. This shipping method would be quite expensive, though, due to the size of the parcel. A more efficient and less expensive method of shipping the desk would be to disassemble the desk into smaller pieces and package it in a much smaller box. The same principle applies to the transmission of digital audio. The cost of sending digital audio is related to its size. The larger the size of the digital audio data, the larger the transmission bandwidth has to be.


Two different compression techniques are typically used in order to squeeze the audio information into a smaller package.  Both compression techniques rely on the fact that audio signals have a great amount of redundancy.  


One commonly used technique is Redundancy Coding. This technique is similar to the process used when computers compress files into ZIP files. This type of compression can be demonstrated by a simple example:

If we need to send the number 1000000, we may compress it to 10⁶.

The compressed number represents the same digits as the original number, but comprises fewer characters. If each character requires four bits, the uncompressed number would require 28 bits (4 bits x 7 characters), while the compressed number would require 12 bits (4 bits x 3 characters). The technique is effective with data where there is lots of redundant information as is the case with digital audio signals. However, the technique has one major drawback; namely that it is relatively time-consuming. It is therefore not very suitable for hearing aid applications where a minimal delay is of the utmost importance.


Another commonly used technique for achieving a low bit rate is Irrelevance Coding.  It is widely used today to create MP3 and other types of digital audio files.  Widex also uses this technique for analysing and compressing audio data in a special part of the codec. .


It is well-known that raw, uncompressed audio contains more information than the human ear can actually detect. Our irrelevance coding algorithm removes all perceptual redundancies by extracting all of the irrelevant audio information which cannot be heard by the listener due to psychoacoustic masking effects in the cochlea. In other words, the irrelevance coding algorithm uses knowledge of masking to remove audio signal elements which are outside the limits of the human auditory system.  


Irrelevance coding relies on a phenomenon known as the Simultaneous Masking Effect. When listening to a soft and a loud sound simultaneously, it is often difficult to hear the soft sound because it is drowned by the loud sound. The masking effect is largest when the soft sound is in the same frequency range as the soft sound (Moore, 2006: 66). This psychoacoustic phenomenon is very useful vis-à-vis audio data compression. The irrelevance algorithm utilizes knowledge of this psychoacoustic effect to remove softer, less dominant sounds which will be masked by louder, more dominant sounds, from the audio signal.


Basically, our irrelevance algorithm reduces the amount of audio data that needs to be transmitted wirelessly by removing sounds which would not be audible to the listener in any case. And by removing sounds in the audio signal which the listener cannot hear anyway, while preserving the sounds the user can hear, the audio signal can be reduced significantly in size without compromising the high sound quality.

Safe digital transmission

Channel coding

An important aspect to consider when sending any type of digital data over a wireless connection is the potential for errors induced by radio frequency interference.


Errors will occur from time to time with any kind of transmission and especially in wireless transmission. The distance between the devices may change, the orientation of the antenna in the controller might be altered, or interference from radio noise might disturb the connection. Such errors must be handled effectively to minimise the inconvenience caused to the hearing aid user in the shape of crackling, dropouts in the sound, etc. Therefore, in order to prepare the audio data for transmission and ensure the integrity of the transmission with respect to correct receiver as well as the quality of transmitted data, channel coding is introduced.

Error detection and handling

The main task of the channel coding algorithm is to provide a method for ensuring that the digital audio signal which is received is indeed correct and error free. A common and very basic way to do this is to calculate what is called a checksum on the basis of the compressed data. The checksum is enclosed in the shipment alongside the data. When the data are received at the other end, a comparison of the checksum and the data is conducted to determine if any errors have been introduced into the digital audio signal.




Figure 2. The data in the shipment are compared
to the checksum on arrival to determine if any
errors have arisen.


However, a checksum will only establish whether or not an error has occurred. It does not provide a solution to how errors should be handled. A major advantage of Widex' channel coding algorithm is that, thanks to a so-called error corrective code, it ensures that a restricted number of errors can be both detected and corrected.

In a Bluetooth transmission technology, for example, the channel coding will by default ask for a retransmission of data which did not pass the receiver's error checking algorithm (www.bluetooth.com). Since retransmission requests will cause an extra delay whenever the system has to wait for the repetitions of the data to arrive, such a method is not a very good choice in a hearing aid application.

Another method involves the removal of audio data packages which contain errors. This method is for instance used in connection with DAB (Digital Audio Broadcasting). DAB-receivers cannot request a retransmission of signals vitiated by errors. Errors in audio data are simply handled by making dropouts in the sound, resulting in there being no sound playback when errors occur. An error handling method which results in clicks or dropouts in the sound is obviously not very suitable for a hearing aid application either.

Widex rely on a special channel-coding technique which is based on the principle of Graceful Degradation. This technique has none of the unfortunate by-products (long delay and dropouts) mentioned above. Instead, it provides a smooth, seamless listening experience for the hearing aid user.



Figure 3. Widex's channel coding algorithm
can both detect and handle errors.


Widex's channel coding algorithm has been designed to ensure that a small number of transmission errors can be corrected by the algorithm itself. A larger number of errors will be handled by means of the above-mentioned Graceful Degradation-based technique. This technique ensures that a large number of errors will not result in any abrupt changes in the output signal, such as dropout or cracking, heard by the hearing aid user. If the errors are too numerous for the algorithm to be able to correct them, the result will be a gradual fading of the sound. While transmission is interrupted, the HA will switch to the Master program. When the quality of the transmission channel is good enough to allow audio transmission once more, the sound will gradually fade in again to provide a nice, smooth listening experience for the hearing aid user.

Figure 4. Summary of three methods for error handling in connection with digitally transmitted data.

Digital Audio Transmission by WidexLink

 

The WidexLink technology makes it possible to minimise the amount of data that needs to be transmitted in order to ensure a high quality output signal. This is essentially possible thanks to the identical audio generators in the encoder and the decoder. More specifically, because the WidexLink encoder and decoder both contain identical synthetic audio generators, it is not necessary to transmit the original signal, or even the synthetic signal. All that needs to be transmitted is information about the discrepancy between the original sound signal and the synthetically generated signal. A more detailed discussion is included in the sections below.


The WidexLink encoding procedure can be divided into five main stages. In the first stage, the original audio signal is compared with a synthetic signal generated by a Synthetic Audio Generator. A Discrepancy Analyser generates information about how close the synthetic sound is to the original. A perceptual model is then applied to determine if discrepancies are audible or not (irrelevance coding). To keep the amount of data to a minimum, only audible discrepancies are allowed to influence the sound generation process. Next, the best approximation to the discrepancy between the original sound and the synthetic signal is retrieved from a large number of synthetic sounds stored in a Sound Sample Archive. And finally, information about which sample is the best approximation to the discrepancy between original and synthetic signal is transmitted to the decoder.

Figure 5. The WidexLink encoder principle.


With WidexLink, the sampling rate is 25.44 kHz, which means that the above procedure is repeated 25,440 times for each second of sound. This enables us to exploit the fact that there is typically not very much variation from one sound sample to the next to generate an increasingly accurate synthetic representation of the original signal.


The first time a discrepancy analysis has been conducted, information about the best approximation is used to create a Sound Model with information about the discrepancy between original and synthetic sound. This knowledge about the discrepancy between synthetic and original sound contained in the model is permitted to influence the generation of the next synthetic sound, whereby the difference between the new synthetic sound and the original sound sample can be reduced. By updating the model every time a sample has been processed by the encoder, the system is able to reduce the difference between original and synthetic signal to a minimum very quickly.


The decoder in the hearing aid contains an exact replica of the synthetic audio generator module in the encoder. Thus, information about the discrepancy-based best approximation is sufficient to provide all the necessary information for the synthetic audio generator in the hearing aid to be able to generate an exact copy of the synthetic signal in the encoder. In other words, the output signal generated by the hearing aid is a 100% synthetic sound identical to the synthetic sound generated in the encoder. The decoding sequence is presented schematically below.




Figure 6. The WidexLink decoder principle.

Transmission Robustness

A new, highly accurate and robust receiver has been developed for the Widex Clear product range to ensure safe transmission and low battery drainage. The accuracy of this new, patent-pending receiver enables us to operate with a low transmission power, which in turn contributes towards extending battery life.

Moreover, the modulation technique employed to send digital data over the wireless system also contributes to maintaing a high degree of robustness.

When sending digital data over a wireless system, a modulation scheme must be used. Modulation essentially determines how the digital information is sent through the radio frequency spectrum. One very commonly used digital modulation technique is Frequency Shift Keying(FSK). In an FSK modulation system, two frequencies are used to represent a binary "0" or a binary "1", respectively (see figure 7 below). So once the audio has been digitised, i.e., turned into a series of 0s and 1s, it can be transmitted by means of two different transmission frequencies which represent either a 0 or a 1. The receiver has to detect which of these two frequencies is being transmitted in order to determine if the transmitter is sending a 0 or a 1. The receiver then demodulates the signal by interpreting the frequencies received as either a 0 or a 1.

Figure 7. Illustration of the Frequency Shift Keying principle. In an FSK modulation system, two frequencies represent a binary "0" or a binary "1", respectively.


Traditional methods of wireless demodulation employ a simple two point sampling of the received wireless signal. This means that, in effect, the receiver relies on only two measurering points when it has to determine if the received signal is a "1" or a "0". Figure 8 below contains a model of the demodulation of a signal without noise.

Figure 8. Model of the demodulation of a signal without noise. Traditional methods of wireless modulation employ two points of sampling. In the example, the sample points indicate a rise. The signal will therefore be interpreted as a 1.

 

The demodulation method described above works very well with a clear signal and no noise. However, as illustrated in figure 9 below, using only two sampling points to determine if the transmitted signal is a 1 or a 0 can result in mistakes if noise is also present in the signal.

Figure 9. Model of the demodulation of a signal with noise. Signals can be wrongly identified when noise is present in the signal.

 

Consequently, Widex has developed a more secure variety of the FSK-method for the reception of wireless transmissions with a high degree of noise in the transmitted signals. The new method introduces a larger number of measuring points than the traditional two, which means that the receiver is able to determine with a much higher degree of certainty whether a received signal should be interpreted as a 0 or a 1.

Application

The new digital transmission technology is a central element in our new Widex CLEAR product range. It is used for the transmission of both data and audio signals in a large number of situations.

Transmission of audio

WidexLink is used for the transmission of audio signals from external devices to the hearing aids when the user watches TV, talks on his mobile phone, or listens to music.

Transmission of Audio via WidexLink
Typical Situation:	Sender and recipient:
TV	TV-Dex – Hearing aids
Hi fi	TV-Dex – Hearing aids
Mobile phone	M-Dex – Hearing aids
Personal audio device (Ipod, mp3 player, etc.)	M-Dex – Hearing aids

 

Transmission of data

WidexLink is also the employed in the transmission of data between remote control (RC-Dex) and hearing aid, and in the exchange of synchronization data between hearing aids (Inter-ear communication).

Transmission of data via WidexLink
Sender and recipient:	Feature:
RC-Dex - hearing aid	Remote control
Hearing aid - hearing aid	HA synchronisation Volume control Program shift
Hearing aid - hearing aid	IE coordination Compression Feedback cancellation Noise reduction
Hearing aid - hearing aid	WidexLink Surveillance Lost partner alarm

 

References:

Moore, B. C. J. (2006). An introduction to the psychology of hearing. Elsevier Publishing Company

www.bluetooth.com

 

 

 

Preserving the fundamentals of a natural sound experience

3D TruSound

Preserving the fundamentals of a natural sound experience

 

InterEar communication through a new, advanced wireless technology

Our new, proprietary WidexLink technology, which has been developed specifically for data exchange and audio streaming in a hearing aid system, offers new and exiting possibilities for exchange of data between hearing aids, and between hearing aids and external devices. The new technology enables the left and right hearing aid in a pair to share the information obtained by the opposite hearing aid, so that information from both ears is taken into account during signal processing. We call this data exchange InterEar communication.

InterEar communication is part of the foundation of the advanced features that comprise 3D TruSound. 3D TruSound is a new dimension of sound processing which aims at preserving the fundamentals of a natural sound experience and providing the highest possible sound quality at the same time.

The TruSound inheritance is excellent sound quality. With the introduction of 3D TruSound, a new dimension is added; namely the preservation of important localisation cues in natural hearing. In addition to the preservation of important sound localisation cues, coordinated noise reduction in difficult listening situations, and enhanced sound quality features are also central elements in 3D TruSounds.

3D TruSound includes the Digital Pinna – a feature which simulates the shadow effect of the outer ear in natural hearing. Furthermore, 3D TruSound also features TruSound Softener which can handle ultra short and extremely fast changes in sound level – for instance when somebody drops cutlery in a metal sink.

Figure 1. The3D TruSound features

 

The 3D TruSound features will be introduced in more detail in the following.

 

The Fundamentals of a Natural Sound Experience

Important sound localisation cues in normal directional hearing

The ability for hearing aid users to determine where sounds are coming from is important for a number of reasons, including safety (e.g. traffic sounds) and communication (e.g. locating a new speaker in a group).

Normal directional hearing relies on the comparison of auditory input from two ears. When an insect flies around our head we are able to determine where it is even with our eyes closed. That is only possible because we have two ears and a brain that help us coordinate the information from both sides of the head.

One of the primary psychoacoustic cues used for localising a sound source to the right or to the left is the split-second delay between the time when a sound reaches the near ear and when it reaches the far ear. This delay is referred to as the interaural time difference (ITD). The ITD will be at its maximum when the sound originates directly from the sides of the head. It is not a large difference; the maximum is around 0.65ms (Plack, 2005). When the sound is coming from the front or the back, the distance from the source to the ears is the same. Thus, there is no interaural time difference for sounds coming directly from the front or the back.

 

Figure 2. Sounds coming from the left or the right will reach the near ear a little sooner than the far ear. The ITD will be at its maximum when the sound comes directly from one side of the head.

 

Another psychoacoustic cue that is used for horizontal localisation is the interaural level difference (ILD). When a sound source is located to the right or the left of the head, the sound will have a greater intensity level when it reaches the near ear than when it reaches the far ear. This difference in sound pressure level at the near and far ear is an important cue for the localisation high-frequency sounds. The effect is predominant for high-frequency sounds because of their short wavelengths. At low frequencies, the difference in level at the two ears from sound coming from the side is low, because long waves easily flow around the head. A head is not a very big object for a low-frequent sound with a long wavelength, but it is a large obstacle for a high-frequency sound with a short wavelength.

Figure 3. Example of the interaural level difference (ILD) at different frequencies measured for one person. Notice the predominance in the high frequencies.

 

It is widely accepted (e.g., Middlebrooks & Green, 1991; Wightman & Kistler, 1992; Schub et al., 2008) that the ILD delivers the primary cue for horizontal localisation of sound sources in the high-frequency region and that the ITD delivers the primary cue for the localisation of low-frequency sounds. In this context, the split between high and low frequencies is approximately 1.5 kHz.

Research (Bogaerts et al., 2006) has shown that wearing a pair of uncoordinated hearing aids can have a destructive effect on the cues used for localisation and consequently reduce the localization abilities of the hearing aid user. However, with the new technology in CLEAR440, it is possible to preserve important psychoacoustic cues.

InterEar TruSound compression – preserving important localisation cues

In a normal compressions system, gain changes depending on the input level. Thus, when you are wearing an uncoordinated set of hearing aids, gain will be prescribed independently to the near and far ear on the basis of the input level at the individual ear. A sound coming from the side will have less intensity when it reaches the far ear than when it reaches the near ear. And because the sound is lower in intensity when reaching the far ear, more gain will be provided at the far ear than at the near ear. This means that the natural interaural level difference will be compromised. However, by coordinating the gain changes on the two sides, the natural ILD can be preserved. This is what the InterEar TruSound Compression does.

Specifically, in CLEAR440 hearing aids, the communication between two coordinated aids ensures that the input levels at the hearing aids are constantly compared (21 times per second), and that the compression response is adjusted accordingly to reflect the difference in input level at the two ears. In practical terms, this means that both ears will receive the same amount of gain, which will depend on the basis of the input level in the near ear.

 

InterEar volume shift and program control – for the preservation of important sound localisation cues and ease of use

The main purpose of coordinated compression is to preserve the ILD. However, if the user turns the volume up or down or changes program at one side only, the ILD can no longer be preserved. In order to prevent that the attempt to preserve the natural ILD is obstructed, InterEar volume shift and program control is introduced in CLEAR440 hearing aids. The InterEar volume shift and program control ensures that if the user changes the volume on one hearing aid the volume of the other hearing aid will also change accordingly. If the users changes program on one hearing aid, the same program will be chosen automatically by the other hearing aid. InterEar volume shift and program control is switched on as default in CLEAR440 hearing aids, but can be switched off by the hearing care professional in the fitting software.

Another major advantage of the InterEar volume shift and program control is that it makes it a lot easier for the hearing aid user to adjust the volume. Users wearing an uncoordinated pair of hearing aids have to adjust each hearing aid separately every time they need to turn the volume up or down or change program. With a pair of coordinated CLEAR440 hearing aids, the user only has to make the adjustment once, which is likely to be appreciated by many hearing aid users.

There may be situations where the hearing aid users would prefer to have different programs in their hearing aids. To accommodate this, a selection of compound packages is available in CLEAR440. These include:

• Master – Telecoil
  
• Master – Microphone+ Telecoil
  
• Master – Reverse focus
  
• Master – Zen
  
• Master – Audibility Extender
  

 

InterEar Speech Enhancer - Coordinated noise reduction in difficult listening situations

Communicating in a noisy environment is one of the most challenging situations for hearing aid users. Many users experience difficulties focusing on one speaker and leaving out the rest. The Speech Enhancer can be very helpful in that situation, and with the introduction of the WidexLink technology the situation can be improved even further.

The Speech Enhancer system is available in Widex high end hearing aids. The system is based on the standardised measure of SII (Speech Intelligibility Index) (ANSI S3.5). The system is unique in that it is able to take the hearing loss into account and optimise speech intelligibility by means of a constant calculation of the SII during noise reduction.

The Speech Enhancer contains a fast-acting mechanism which is able to add gain to frequency areas with speech to further optimise speech intelligibility. In CLEAR440 hearing aids, this fast-acting mechanism is coordinated to ensure that it is active on the side with the most dominant speaker. By exchanging important percentile data over the WidexLink, two CLEAR440 hearing aids are able to make a decision on whether to activate the fast-acting gain application mechanism and on which side to do this. This way, the Speech Enhancer systems will no longer base its decision to act on one-sided data input, but on data input on the sound environment on both sides of the head. Only on the side where speech is dominant will the speech enhancer actively work to preserve speech audibility by adding gain to frequency regions important for speech. On the opposite side the noise reduction system will work to keep the noise below the threshold of the listener using his or her hearing threshold data in the calculation. In practical terms, the coordination between hearing aids means that in a cocktail party situation with many speakers, the InterEar Speech Enhancer in CLEAR440 supports the singling out of the dominant speaker.

Digital Pinna – re-creating the natural pinna shadow effect

The ear has some natural directional characteristics, mainly due to the physical presence and shape of the pinna. One example is the pinna shadow effect. Sounds coming from the front reach the ear canal almost directly, whereas sounds coming from behind are obstructed by the pinna and are thus somewhat attenuated before they reach the ear canal. This pinna shadow effect mainly affects the region around 2 to 5 kHz, where sounds coming from behind are attenuated by 3-4 dB relative to sounds coming from the front. This natural 3-4 dB difference is an important cue for the listener to know whether a source is in front or behind.

Microphone location is known to have a negative impact on the ability to determine if a sound is coming from in front or from behind. Front-back localisation is especially a problem for users with behind-the-ear (BTE) hearing aids, because the location of the microphone essentially offsets the normal localisation cues provided by the pinna (the outer ear).

For example, Westermann and Tøpholm (1985) found that BTEs give poorer localisation than ITEs, in particular with respect to front-back confusions. A BTE hearing aid captures the sound at the position of its microphone(s), i.e. above and behind the pinna. This means that the pinna shadow effect is not preserved in the signal that the HA provides, and consequently the HA user loses some of the ability to localise vertically, and to distinguish front and back.

 

Figure 4. The average pinna effect measured over 45 heads. The 3-4 dB difference between sound coming from behind and from the front is seen in the 2-5 kHz frequency region.

 

As mentioned, the spectral shaping provided by the pinna is an important cue for front/back localisation. One way to improve localisation for hearing aid users with BTEs, therefore, would be to attempt to re-create this shaping through processing of the input signal. This is precisely what Digital Pinna in CLEAR440 does.

A series of developmental experiments have shown that the pinna shadow effect can be simulated by introducing a restriction on the adaptive locator. More specifically, the natural attenuation of sounds coming from behind can be re-created by setting the frequency bands from 2 kHz and up (bands 10-15) in fixed directional mode (i.e. in the hypercardioid pattern, which picks up sound at the front and eliminates most sound from the sides and rear), while leaving the lower bands (1-9) in omni-directional mode.

Figure 5. The microphone patterns of Digital Pinna. The lower bands (1-9) are in omni-directional mode, while the upper bands (10-15) are in fixed directional mode (i.e., in hypercardioid).

 

The developmental experiments were carried out using four normal-hearing listeners as subjects. 12 loudspeakers were set up in a 360° horizontal circle around the test subject, with a radius of 1.2 meters and 30° between the loudspeakers. The setup is shown in the figure below.

Figure 6. The loudspeaker setup in the developmental experiments exploring Digital Pinna.

 

As stimuli, 4 different recordings of a wooden rod (120cm) that is dropped on the floor were used. Each of these 4 recordings was presented once from each of the 12 loudspeakers, giving a total of 4*12=48 presentations (in randomised order) for each localisation test. The test subjects were instructed to indicate from which loudspeaker each presentation was coming. The Digital Pinna was compared with the omni-directional mode and with a fixed locator (the one giving maximum directivity).

Figure 7 below shows the main results for the four different tests where head movements were not allowed. Digital Pinna resolved 80% of the front-back confusions that were made in omni-directional mode. This performance is slightly better than (or at least similar to) the fixed directional system. In addition, Digital Pinna did not degrade the horizontal localisation as the fixed directional system did, and Digital Pinna had the best horizontal localisation in the frontal plane.

 

 

Figure 7. Main result of a developmental experiment comparing front-back confusion with Digital Pinna, omni-directional and fixed directional mode. Head movements were not allowed in this condition. The left panel shows how many front/back confusions were made (in %) in each mode. The right panel shows by how many degrees horizontal (left-right) localisation errors deviated from the correct response angle on average.

 

When head movements were allowed, the Digital Pinna was clearly best overall. The results are displayed in figure 8 below. There were almost no front-back confusions with Digital Pinna compared to omni-directional and fixed directional mode, and the horizontal localisation did not show large errors in the back, like the fixed directional mode did.

 

Figure 8.
Main result of a developmental experiment comparing front-back confusion with Digital Pinna, omni-directional and fixed directional mode when head movements were allowed. The left panel shows how many front/back confusions were made (in %) in each mode. The right panel shows by how many degrees horizontal (left-right) localisation errors deviated from the correct response angle on average.

 

 

The results from the developmental studies indicate that Digital Pinna restores the ear's natural effect (pinna shadow) in BTEs, and thus the user's ability to distinguish between sources in front and back. The results also showed that it did so without degrading horizontal localisation (which the fixed directional system may). Furthermore, it enables the user to improve localisation by taking advantage of active head movements (which the fixed directional system does not).

Importantly, the microphone system remains adaptive when Digital Pinna has been activated. In quiet listening environments, the lower bands (1-9) will be in omni-directional mode, whereas the upper bands (10-15) will be in hypercardioid as shown in figure 5. In noisy listening conditions, the directionality of both lower and upper bands will increase to yield optimum speech intelligibility. The microphone mode of the lower bands may adapt any characteristic from omni-directional to bipolar, while the upper bands may move from hypercardioid towards bipolar as the signal-to-noise ratio worsens.

 

Advanced Sound Quality Features - Taking high sound quality a step further

A discussed above, one of the key elements in the CLEAR440 hearing aid is a collection of features designed to preserve important psychoacoustic cues. Another cornerstone in the CLEAR440 product family is a collection of enhanced sound quality features. These features will be described in more detail in the following.

InterEar feedback cancelling

To be successful, a feedback system must be effective in terms of eliminating feedback. The precision with which it can determine if the signal really is a feedback signal is also important.

Experience has shown that the Multi-directional active feedback cancelling system is extremely efficient in controlling dynamic feedback problems. No matter if the hearing aid user is talking on the phone, hugging a friend, or putting on a hat, the Multi-directional active feedback cancelling system has been designed to ensure that whistling does not occur.

With the introduction of CLEAR440, we have managed to make our industry leading system even more precise. Specifically, when an external, autocorrelated sound like a whistle or an alarm is picked up by the hearing aid, the InterEar coordination between the hearing aids means that they are able to compare detected sound from both sides of the head. If the feedback-like signal is the same on both sides, it can be deducted that it is an external sound rather than a feedback signal which has been detected. Thus, with InterEar feedback cancellation, we are able to avoid gain regulation when it is not necessary as a result of "false positives". However, if a feedback-like sound is only found on one side of the head, the system will deduct that it is feedback which needs to be handled.

 

Enhanced bandwidth

Audio bandwidth is one of the key factors in maintaining a high sound quality. Thanks to the new technology in the CLEAR440 product range, we are able to offer an exceptionally broad audio bandwidth in models with a Clearband receiver, stretching from 70 Hz to 10.5 kHz in the music program, and 100 Hz to 11.2 kHz for digitally transmitted sound. This is industry leading.

One area where a broad audio bandwidth makes a clear difference for hearing aid users is when they listen to music. The high frequencies provide ambience and brilliance to the sound. Thus, the sound experience will be somewhat richer with an upper bandwidth of 10.5 kHz when listening to a hi-hat or cymbals, for instance. Similarly, an audio bandwidth stretching as far down as 70 Hz will produce a fuller bass.

 

TruSound Softener

The advanced sound quality features of 3D TruSound also include the TruSound Softener. The purpose of the TruSound Softener feature is to make impulse sounds, such as rattling porcelain or hammer blows, less annoying without removing them from the surrounding sound environment or making them unnaturally soft. The TruSound Softener is described in more detail in a separate whitepaper entitled TruSound Softener: A new algorithm for detecting and handling impulse sounds.

 

Summary

With 3D TruSound, Widex takes sound processing a step further. 3D TruSound introduces a new dimension in sound processing – the preservation of the fundamentals of a natural sound experience. This is achieved by taking signal analysis data from the opposite hearing aid into account in the processing.

The new proprietary WidexLink technology in the CLEAR440 product family makes it possible to preserve a number of important psychoacoustic cues used for determining where a sound is coming from.

Our attempt to provide a realistic listening experience for the hearing aid user has lead to the development of a comprehensive collection of features, including

• Digital Pinna, which has been developed to support front-back localisation
  
• InterEar TruSound compression, and InterEar volume shift and program control, which contribute to the preservation of psychoacoustic cues used for localising sound coming from the sides.
  
• InterEar Speech Enhancer, which may help hearing aid users focus on the dominant speaker in noisy situations
  

Another cornerstone in the CLEAR440 product family is a collection of enhanced sound quality features. The result of our latest effort to provide the best possible sound quality include

• The TruSound Softener, which has been designed to detect and handle impulse sounds
  
• Extended bandwidth in both high and low frequencies
  
• InterEar feedback cancelling which minimises the risk of continuous steady sounds being attenuated by the feedback system because they are mistaken for feedback
  

  
   

 

References

ANSI S3.5. 1997. American National Standard: Methods for the calculation of the Speech Intelligibility Index.

Algazi, V. R.; Duda, R. O.; Thompson, D. M. & Avendano, C. (2001). The CIPIC HRTF Database.
In proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 99-102.

Middlebrooks, J. C., & Green, D. M. (1991). Sound localization by human listeners. Annual Review of Psychology, 42, 135-159.

Plack, C. J. (2005). The sense of hearing. New Jersey: Lawrence Erlbaum Associates

Schub, D. E., Carr, S. P., Kong, Y., & Colburn, H. S. (2008). Discrimination and identification of azimuth using spectral shape. Journal of the Acoustical Society of America, 124(5), 3132-3141.

Van den Bogaert, T., Klasen, T., Moonen, M., Van Deun, L., & Wouters, J. (2006). Horizontal localization with bilateral hearing devices: Without is better than with. Journal of the Acoustical Society of America, 119(1), 515-526.

Westermann, S. & Tøpholm, J. (1985). Comparing BTEs and ITEs for localizing speech. Hearing Instruments, 36(2), 20-24.

Wightman, F. L. & Kistler, D. J. (1992). The dominant role of low-frequency interaural time differences in sound localisation. Journal of the Acoustical Society of America, 91(3), 1648-1661.

Clear 440 Launch

The Clear 440 platform was launched very successfully to the UK & Irish market last weekend at the AIHHP conference at the Cobham Hilton in Surrey. The platform was well received by all who attended, with particular interest paid to the interear features and strategies and also the wireless connectivity.

We have launched initially the RIC format and BTEs and towards the end of the year, perhaps early next year  ‘in the ear’ versions will be released. The general feeling was that the connectivity options  look really elegant . Initially there will be options to stream wirelessly from TV and audio sources (TV-Dex) with no discernable time lapse or lip sync issues, the mobile phone streaming option (M-Dex) will follow very shortly. The new remote control option for the Clear 440 (RC-Dex) is very stylish, simple and intuitive.



Features in the Widex Clear 440 include Binaural Synchronisation, the new Widex range of hearing aids will ‘talk’ to each other in binaural fittings, this will  lead to better localisation, speech enhancement and improvements in hearing speech in noise. Another benefit is great feedback management and significantly improved directionality of sound.

Another couple of interesting features built into the new Clear 440 are an enhanced bandwidth , in fact the widest bandwidth currently available including enhanced low frequencies down to 70Hz, good for full appreciation of music. Also FreeFocus, allowing a Patient to hear in selected directions without turning their head.

We will begin to post educational posts pertaining to the Clear platform, features, strategies and fitting on the blog shortly.

Using the Program manager in mind440 fittings

This bulletin describes how to use the Program manager in mind440 fittings using Compass V4.5. The descriptions refer to the Program manager in the Fine tuning window. You access the Program manager from the Toptool bar. Please refer to Compass in order to see the details of the screen. Remember that Compass offers several help functions to explain the different options in the program: Tooltips, Solution guide, Using this window panels and the help manual.

The mind440 hearing aid is, by default, a one program hearing aid, with a Master program designed to provide the best possible overall performance in all environments with all adaptive features working together to optimise audibility, intelligibility and comfort. However, some hearing aid users encounter environments in which they may wish the hearing aid to operate with a different focus. For this purpose, mind440 features a range of additional listening programs that can be selected initially during the fitting procedure, or later on during follow-up.In the Fine tuning window, the Program Manager is accessible from the Toptool bar. The Program manager allows you to select, copy and delete listening programs, as well as to change the order of the programs. The Program manager offers a selection of different program templates that may serve as inspiration in the hearing aid fitting process.

  The Program manager allows you to select, add, copy, delete and change the order of the programs in the hearing aid

Using the Program manager
To add a program, select the desired program name in the program list and select the add arrow button or double click on the selected program. Whenever you point to or select a program, an illustration or a short explanation of it is displayed on the screen. Click the button in the lower right hand corner to shift between illustration and explanation. The Program manager allows you to select more than one program of the same type either by selecting two identical programs from the program list or by using the copy button in the program item you want to copy. Selecting identical programs from the program list gives you programs with default
settings, while a copy will include any fine tuning performed in the original program. You can change the program order by pressing the arrow buttons in the program item in the Program manager.

Press the delete (trash can) button for a program item you want to delete. Note that the original Master program can never be deleted. Before you delete a program, it can often be useful to save your fitting in the NOAH database using the save button in the Toolbar. This will allow you to re-establish the deleted program including any fine tuning you may have performed. In binaural fittings you will typically have identical programs in the two hearing aids. It is possible to have a different number and different types of programs in the two hearing aids. Press the appropriate binaural adjustment icon, which allows you to select programs individually for the two hearing aids. Be aware, though, that such a choice requires extra careful counselling of the user. We strongly recommend that the number of listening programs in binaural fittings is identical to avoid any confusion. You should also be aware that two hearing aids cannot be operated with the same remote control unless the number of programs in the hearing aids is identical.

Program types
The Program manager offers a selection of listening programs. A maximum of five programs can be selected in the basic Program manager. Additionally the SmartToggle feature can be added. Note that the number and type of listening programs should always be selected based on an analysis of your client’s auditory needs as well as his/her ability to understand and manipulate the program button or remote control.
Depending on the Direct Audio Input system you use, the M-DAI and DAI programs may be placed at program location 1 and 2. You must use the DAI Manager to set up the DAI system correctly for the equipment you use.

Please be aware that telecoil programs are not available for CIC hearing aids, m models and for ITE instruments without a telecoil.

Master
Provides the best possible overall performance in all environments. All adaptive features work together to optimise audibility, intelligibility and comfort. The default settings in this template are:

Gain/compression characteristic optimised for audibility, intelligibility and comfort
Microphone mode is set to Hd Locator with Speech Tracer (Hd Locator omni for CIC)
Speech and noise mode is set to Speech Enhancer
Feedback cancelling mode is set to SuperGain

Audibility Extender
Provides audibility for high-frequency sounds by means of frequency transposition. This is a program template for users with little or no hearing in the high frequencies. The default settings in this template are:

Linear transposition of high frequency sounds. The start band frequency is defined by the Sensogram configuration.
Microphone mode is set to Hd Locator with Speech Tracer (Hd Locator omni for CIC)
Speech and noise mode is set to Widex classic noise reduction
Feedback cancelling mode is set to Off

Zen
A program playing soothing and relaxing tones by means of fractal technology. The tones are randomly generated, meaning that the tone sequences are never repeated. The Zen style to be played in the Zen program can be selected amongst five different Zen styles. In the default setting the Zen tones are played linked to the Master program. Therefore the acoustic settings are the same in both the Zen and Master programs. The default settings in this template are:

Gain/compression characteristic optimised for audibility, intelligibility and comfort
Microphone mode is set to Hd Locator with Speech Tracer (Hd Locator omni for CIC)
Speech and noise mode is set to Speech Enhancer
Feedback cancelling mode is set to SuperGain

Music
Focuses on optimum reproduction of music, whether it is performed live or played from stereo or surround systems. The program emphasises low frequencies. The default settings in this template are:

Gain/compression characteristic optimised for music (low frequency emphasis)
Microphone mode is set to Hd Locator omni
Speech and noise mode is set to Off
Feedback cancelling mode is set to SuperGain music

TV
Focuses on optimum reproduction of sound from television sets. TV and radio signals are very different from normal acoustic signals produced in the environment, as they have already been processed (compressed). The TV program takes this into account in its compression characteristic. The program attenuates low frequencies and emphasises high frequencies. The default settings in this template are:

Gain/compression characteristic optimised for TV sound
Microphone mode is set to Hd Locator with Speech Tracer (Hd Locator omni for CIC)
Speech and noise mode is set to Speech Enhancer
Feedback cancelling mode is set to SuperGain music

Comfort
Focuses on optimum comfort in noisy environments as well as in quiet environments. The default settings in this template are:

Gain/compression characteristic optimised for comfort (higher knee-point for soft input levels)
Microphone mode is set to Hd Locator with Speech Tracer (Hd Locator omni for CIC)
Speech and noise mode is set to Widex classic noise reduction comfort
Feedback cancelling mode is set to SuperGain

Acclimatisation
This template is designed for the first-time user who needs an alternative setting during the acclimatisation period. It is sometimes an advantage to reduce gain a little, so that the new sound picture does  not cause any discomfort. The gain can then be gradually increased over four acclimatisation stages. The default settings in this template are:

Gain/compression characteristic optimised for first-time users (reduced gain for all input levels (especially in the high frequencies) compared to the Master program)
Microphone mode is set to Hd Locator with Speech Tracer (Hd Locator omni for CIC)
Speech and noise mode is set to Speech Enhancer
Feedback cancelling mode is set to SuperGain

MT
An acoustic program combined with the input from the telecoil. Not available for the m model, CIC and ITE without telecoil. The default settings in this template are:

Gain/compression characteristic set as the Master program
Microphone mode is set to Hd Locator omni
Speech and noise mode is set to Speech Enhancer
Feedback cancelling mode is set to SuperGain

T
Focuses on the input from the telecoil. Not available for the m model, CIC aids and ITE aids without telecoil.

The default settings in this template are:
Gain/compression characteristic set as in the Master program
Speech and noise mode is set to Speech Enhancer

SmartToggle
This feature gives you access to the Zen+ program and is activated by means of a long press on the program up button on the remote control or on the hearing aid program button. A new long key press brings you back to the Master program

The Zen+ program.

The Zen+ program accessed via the SmartToggle feature can contain three different Zen styles. The Zen tones are by default played with the microphone active and linked to the Master program. In Settings, the microphone can be deactivated for each of the Zen styles individually to listen to Zen tones only. Once the SmartToggle feature has been activated, you can use short key presses to toggle between the three different Zen styles.
The Program manager with the SmartToggle feature (Zen+) activated

View program access
The view program access function allows you to view the number of programs available in the Program manager for the following conditions: Without remote control (RC), with RC4-1 and with RC4-2.

The View program access function lets you know the number of programs available in the Program manager

How to demonstrate the different listening programs
Note that the purpose of the Program manager is to select a program combination - not to demonstrate the programs. When the Program manager is open, the Master program will be active in the hearing aid. Once you have closed the Program manager, you can shift between the selected programs using the Program starter, which allows the user to listen to the differences in sound processing.

Changing the feature settings in mind440 fittings using Compass V4.5 and above

This bulletin describes the features you can change and how to do it using Compass V4.5 and above. The descriptions refer to the feature setting options in the Fine Tuning section. The feature setting may be changed for the microphone mode, the speech and noise mode and the feedback cancelling mode. Please refer to Compass in order to see the details of the screen. Remember that Compass offers several help functions to explain the different options in the program: Tooltips, Solution guide, Using this window panels and the help manual.

The feature settings may be changed individually for each of the listening programs in mind440. Such changes are easily made in the Feature settings panel in the Fine tuning section in Compass. In this panel you can change the mode in which the microphone works, the mode in which speech and noise is handled in the hearing aid and the mode in which acoustic feedback is handled in the hearing aid.
The feature settings panel in the Fine tuning window of Compass

Microphone mode
The Microphone mode can be set to one of three settings: The default setting in the Master program is Hd Locator with Speech Tracer. The High definition Locator is a 15-channel fully adaptive directional microphone system that exploits frequency-specific information about the listening environment to improve the signal-to-noise ratio. The Speech Tracer ensures full audibility of single speakers in relatively quiet rooms, regardless of the intensity level and the azimuth of the speaker. The other two options are Hd Locator dir (fixed directional characteristic) and Hd Locator omni (fixed omnidirectional characteristic). We recommend that these two modes are primarily used for demonstration purposes.
The microphone mode options

Speech and noise mode
The Speech and noise mode can be set to one of six settings. The default setting in the Master program is Speech Enhancer. This is a signal processing scheme that utilises information about the individual hearing loss along with current information on the spectrum of the speaker and the noise source to optimise intelligibility according to the Speech Intelligibility Index (SII). You can also select
a classic Widex noise reduction scheme, Noise reduction, in the drop-down list. This would be an option for clients who have previous successful experience with Widex hearing aids and our classic approach to noise reduction. These two settings are supplemented by a less and a more active version, Noise reduction minimal and Noise reduction enhanced, and a version with special focus on listening
comfort in all types and degrees of background noise, Noise reduction comfort. Finally you can choose to deactivate the Speech and noise mode altogether using the setting Off.
The speech and noise mode options

Feedback cancelling modes
The feedback cancelling mode options

The mind440 Multidirectional active feedback cancelling feature integrates knowledge of the acoustic properties of the hearing aid in the ear with directional sensitivity for each of the 15 frequency channels in mind440. Its mode of operation can be set to one of four settings. The default setting of the Feedback cancelling mode in the Master program is SuperGain, meaning that the hearing aid gives as much gain as possible without the risk of whistling. You can use the drop-down list to change the setting to SuperGain music, especially designed for listening to music, or to SuperGain max, designed to give as much gain as possible, even if you risk that the hearing aid occasionally whistles. You also have the option to turn the feedback cancelling system off altogether, in which case an extra adjustment parameter will appear. This parameter allows you to adjust the static max gain limit across frequencies in case of whistling. Deactivating the Multi-directional feedback cancelling feature is only recommended in fittings with considerable headroom in the maximum available gain without feedback. Otherwise recurring acoustic feedback may be a considerable problem for your client.

PDF Copy of the bulletin for download Changing feature settings

The Audibility Extender and how to individualise it

This bulletin describes the Audibility Extender program, the default settings in Compass, how to fine tune the Audibility Extender program as well as how to individualise it. The descriptions in this bulletin refer to the Graphics panel and the Audibility Extender settings panel in the Fine tuning window of the program in  Compass. Please refer to Compass in order to see the details of the screen. References will be found in the final section.

Audibility Extender default setting in Compass 

The Audibility Extender program uses linear frequency transposition and is developed for clients with little or no hearing in the high frequency area. Compass uses Sensogram data to set the start frequency of the transposed signals where normal aided speech cannot be made audible by normal amplification. The frequency range of the transposed signals is, by default, set to three frequency bands.



In the Fine tuning window in Compass you can see how the start frequency is automatically set, based on the Sensogram.

Fine tuning the Audibility Extender program

In the Fine tuning window, you can fine tune the Audibility Extender program. The elements that can be changed are:
The start frequency o • f the transposed signal
• The loudness of the transposed signal
• The frequency range of the transposed signal (Basic or Expanded)

Individualising the Audibility Extender program

Proper fine tuning ensures that the client is able to detect and discriminate relevant sounds. If a client
fitted with the Audibility Extender in the default setting has not benefitted from the program after a period of acclimatisation, it is recommendable to fine tune the start frequency and the Audibility Extender gain. One way to find the optimal setting is to make sure that the hearing aid user can hear the/s/ sound at normal speech level (approx. 30 dB HL). The energy of the /s/ sound lies in the range between 4 KHz and 6 KHz and is of great importance in speech understanding in many languages. It should therefore be in the transposed signal and be audible with the appropriate settings.

A simple method, based on this, is described below:

1. 
   
   Increase The start frequency three steps


2. 
   
   Check the audibility of the /s/. If audible end, if not audible proceed


3. 
   
   Adjust AE gain in steps


4. 
   
   Check the audibility of the /s/ after each step. If audible end, if not audible proceed


5. 
   
   Lower start frequency one step (reset AE gain to 0


6. 
   
   Repeat from step 2       

For clients who cannot hear the /s/ sound even at lowest start frequency, use the /sh/ sound (the energy
of this sound lies in the range from 2 KHz to 4 KHz). Be aware that it will normally take some time for the client to get used to the new sound. Several clinical trials have shown that benefit of the Audibility Extender improves over time (Kuk et al. 2008)

Frequently asked questions:

Who are candidates for the Audibility Extender?
Candidates for the Audibility Extender are both children and adults with an unaidable high frequency hearing loss. Clients who have hearing loss that is greater than 70 dB at and above the start frequency can often benefit from the Audibility Extender.

Verified fitting range for the Audibility Extender. From Kuk et al 2008

Is fine tuning always necessary?
If the client is satisfied with the Audibility Extender in the default setting, fine tuning is not necessary. If the client cannot get used to the sound in the Audibility Extender program after an acclimatisation period, fine tuning is necessary.

How long is the acclimatisation period in general?
Studies indicate that clients still improve in a speech discrimination test 6 weeks after being fitted with the Audibility Extender (Auriemmo et al 2008, Kuk et al 2007).

Can I make an objective evaluation of the settings of the Audibility Extender?
It is possible to make an objective evaluation of the settings in the Audibility Extender program by using SoundTracker: Say the /s/ sound and look at the bars of the transposed signals in SoundTracker. If the bars of the transposed signals exceed the thresholds, the sound should be audible. This is also a method that can be used if the client cannot provide a verbal response.

References and other relevant literature

Auriemmo, J., Kuk, K., & Stenger, P. (2008). Criteria for evaluating the performance of linear frequency
transposition in children. Hearing Journal, 61(4), 50, 51-54
Auriemmo, J., Thiele, N., Marshall, S., Quick, D., Pikora, M., & Strenger, P. (2008). Effect of linear
frequency transposition in school-aged children. AAA. American Academy of Audiology 2008, 1
Kuk, F, Keenan, D., Peeters, H., Korhonen, P., & Auriemmo, J. (2008). 12 Lessons learned about linear
frequency transposition. Hearing Review, 15(12), 32, 34, 36-38, 40-41
Kuk, F. (2007). Critical factors in ensuring efficacy of frequency transposition. Part 1: Individualizing
the start frequency. Hearing Review, 14(3), 60, 62-64, 66
Kuk, F., Keenan, D., Peeters, H., Lau, C., & Crose, B. (2007). Critical factors in ensuring efficacy of
frequency transposition part 2: Facilitating initial adjustment. Hearing Review, 14(4), 90, 92, 95-96.
Kuk, F., Korhonen, P., Peeters, H., Keenan, D., Jessen, A., & Andersen, H. (2006). Linear frequency
transposition: Extending the audibility of high-frequency information. Hearing Review, 13(11), 42,
44-46, 48.

Fitting mind330

Fitting mind330
mind330 is a mid-range product in the mind™ product family. It contains a number of the features
included in our top-range product mind440, and the fitting bulletins for mind440 apply to mind330
in most cases. However, not all features included in mind440 are available in mind330.

This bulletin outlines the most important differences between the mind330 and mind440 series and
describes the areas where the fitting of mind330 products differs from the fitting of mind440 products.
A few changes which will apply to the entire product range in the mind family with the introduction
of Compass V4.7 are also mentioned. The most important differences between the features included in the mind330 and the mind440 series are summarised in the table below:

 

Features which differ in the mind440 and mind330 series

Mind 440

Mind 330

15 channels and 15 bands                                   15 channels regulate compression during processing         The frequency range is divided into 15 bands	10 channels and 10 bands 10 channels regulate compression during processing                                                          The frequency range is divided into 10 bands
Speech Enhancer In the new and improved system, speech perception in noisy environments is optimised even further by combining statistical analyses with information on the client’s individual hearing loss.	Noise reduction The Classic noise reduction system continuously analyses incoming sound statistically in order to distinguish between speech and background noise and minimise the effect of noise.
5 listening programs can be included 5 different programs can be added to the hearing aid from a selection of listening programs.	4 listening programs can be included 4 different programs can be added to the hearing aid from a selection of listening programs.
Zen fractal generator The Zen program with a choice of relaxing tones and chimes is available	– The Zen/Zen+ programs are not available in the mind330 series

The practical implications of the different features in mind330 and mind440 for the fitting process are
described in more detail on the next page.

The Fitting section
In the mind330 series, signal processing takes place in 10 channels, and the frequency range is divided
into 10 bands during analysis. Accordingly, during the feedback test the maximum available gain is displayed across 10 channels, and the noise level panel shows 10 bars.

Maximum available gain is shown across 10 channels. Noise level is displayed across 10 bars.
In-situ thresholds are measured in either 4 bands (Basic Sensogram) or up to 14 bands (Expanded
Sensogram). Because Sensogram measuring bands are independent of the hearing aid’s signal
processing channels, even a 10 channel device such as mind330 can utilise the expanded Sensogram
option to obtain detailed information on the hearing loss configuration across frequencies to ensure
the best possible basis for calculating the gain and compression characteristics.

The Expanded Sensogram can contain up to 14 bands because Sensogram measuring bands are independent of the signal processing channels.

In the Program selection window, four basic listening programs can be enabled for hearing aids in the
mind330 series.

mind330 hearing aids can contain four basic listening programs. The Master program is the only obligatory program. The other programs can be added from the list according to the client’s needs and wishes.

The Fine tuning section

Fine tuning the Master program

The four SoundTracker views, which show the real-time performance of the hearing aid in the current
environment from different perspectives, are shown across 10 channels when a mind330 hearing aid
is being fitted. Similarly, the feedback cancelling system is active in 10 channels in the mind330 series.
The default feedback cancelling mode SuperGain is shown in the example below.
The Loudness master handle allows you to adjust gain for all input levels simultaneously, or for the
low, mid or high frequencies separately as in the example below.
SoundTracker view in 10 channels. Separate adjustment of gain in low, mid and high frequencies. The default
feedback cancelling mode is SuperGain.

Fine tuning the Audibility Extender
When fine tuning the Audibility Extender in a mind330 hearing aid, the start frequency is automatically
set to 2,000 or 4,000 Hz on the basis of the configuration of the Sensogram. You can change the
start frequency if necessary by selecting the desired frequency from the drop-down list.
8

The MPO manager
With the introduction of Compass V4.7, band-specific adjustment of the Maximum Power Output
settings of the four basic frequency bands becomes possible for mind440 and mind330 products. The
introduction of band-specific output adjustment in the MPO lock means that the maximum output
can be increased or decreased in a specific frequency region without affecting the other regions. A
typical situation where this can be useful is when your client has a highly restricted dynamic range
accompanied by a high intolerance to loud sounds in a particular frequency region owing to, for example,
a steep hearing loss in the high-frequency region.
    Band-specific adjustment of the high-frequency region (2,000-4,000 Hz) using the MPO lock.

The Solution guide
From version 4.7 of Compass, we introduce pictograms in the Solution guide to provide a better
overall view of where changes will apply. Specifically, pictograms are displayed for all those programs
which will be affected by the adjustments when implemented.
If you use the Global solution guide to deal with the general performance of the hearing aid, the Master
program and any additional listening programs which are linked with the Master program will be
affected by your adjustments. Pictograms representing the Master program and the programs which
are linked to it will be displayed to remind you that your changes will affect all the programs unless
you choose to unlink some of them from the Master program.
Pictograms indicating that the Master program, Audibility Extender, Comfort program and MT program will be
influenced by the adjustments made via the Global solution guide.

If you use the Program specific solution guide to solve user complaints related to a specific program
other than the Master program, only one pictogram representing that program will typically be displayed
to highlight the fact that the adjustments will apply to that program only. However, the telecoil
program and the microphone and telecoil program may be linked to other programs than the Master
program. In such cases, a pictogram representing the T or the TM program will also be visible to
remind you that your changes will apply to that program also.
Typically, a single pictogram is displayed when the Program specific solution guide is used.

The T and TM programs may be linked to other programs than the Master program. In the example, the TM program
is linked to the TV program and will be affected by whatever changes are made to the TV program