Patent Grant
6876750
This invention relates to digital hearing aids, and more particularly relates to tuning such digital hearing aids to compensate for an individual hearing loss.
Over the last two decades, almost every type of audio equipment turned to digital circuitry to improve and/or enhance performance. Available equipment to assist the hearing impaired (both severe and modest) includes both analog and digital, although digital approaches are gaining more and more ground because of size and circuit flexibility.
For typical loudspeakers, to render high quality audio, inherent variations with frequency in the amplitude, or sound level, of the sound reproduced by the speaker for a given level of signal driving the loudspeaker must be normalized. This process is known as speaker equalization. Traditionally, the design of equalizers has been performed by an experienced technician who uses precision instruments to measure the speaker characteristics and adjusts filters as needed to equalize the speaker. However, as will be appreciated, the speaker may be equalized without taking the environment into account if the speaker will be moved to several different environments, or there is no way to know what the environment might be or the “environment” changes. Alternatively, the speaker can be equalized to account not only for deficiencies of the speaker itself, but also the effects of the environment. In this way, the spectral performance of the loudspeaker is compensated so that for a given audio signal power level the amplitude of the resulting sound is approximately the same for all audio frequencies in the performance range of the loudspeaker. Such a procedure is manual, time-consuming and requires significant expertise but still does not necessarily yield the best equalization possible for the resources expended. In addition, the speaker equalization equipment often requires many cubic feet of space.
Hearing aids, although including an amplifier and a speaker, also include features which are at odds with the typical concert or public address loudspeaker. For example, whereas an amplifier and speaker system used for concerts and the performing arts demands extreme amplification and huge speakers, the smaller the size of the hearing aid the better. Also as was discussed above, where speaker equalization or a flat frequency response of the speaker or sound system is demanded for concerts and public performance, a properly tuned hearing aid does just the opposite. That is, the properly tuned hearing aid carefully avoids amplification of those frequencies at which an individual has acceptable or normal hearing while at the same time providing substantial amplification to frequencies at which the individual is impaired. It is, of course, possible that an individual's hearing loss could be the same across the audio spectrum in which case a hearing aid with a flat response might be desirable. Typically, however, hearing loss is frequency dependent, and for most individuals, the loss is progressively worse at frequencies at the high end of the audio spectrum. Although these differences result in opposite demands for a public sound system and an individual's hearing aid, much of the technical theory required to satisfy these opposite demands is the same. For example, whereas sound system equalization schemes may be used to compensate for sound power or volume output variations at different frequencies to obtain a flat sound system response, the same scheme may be used for sound system “unequalization.” That is, the scheme intentionally varies the power output of the hearing aid at various frequencies to achieve an output which is intentionally not equalized. This compensates for the wearer's impairment so that in most applications, the wearer perceives or hears as flat a response as is possible over the audio spectrum. However, as will be discussed hereinafter, a “hearing aid” may also be used by an individual with normal hearing for purposes of “enhancing” the individual's hearing ability above normal with respect to specific sounds, frequencies or environments. For these uses, the “hearing aid” will not strive to provide the user with a “flat” response over all frequency bands, but may instead intentionally peak the hearing ability at selected frequencies. Consequently, it will be appreciated by one skilled in the art that schemes and discussions related to equalization of individual sound systems are equally applicable to the “unequalization” required by individual hearing aids. Therefore, although much of the following discussion refers to equalization of sound systems by selective frequency dependant amplification, a person skilled in the art will recognize that the technology for achieving frequency dependent amplification to achieve “unequalization” as is necessary for tuning hearing aids is the same.
More recently, automated equalization schemes have been proposed. For example, one such proposed scheme is an automated graphic equalizer. Such an equalizer has a plurality of channels having fixed center frequencies and fixed Qs (ratio of center frequency to bandwidth of the channel) that cover the entire audio band with filters. It has been proposed to automate the equalization process with such an equalizer by using instruments to record the spectral behavior of a loudspeaker in an environment, and then, in an automated fashion apply to varying degrees such filters so as to compensate the loudspeaker performance and thus bring the resulting spectral behavior of the loudspeaker more closely to a target curve. The approach is limited in its capacity for optimization, and the equalizer is complex, making this approach impractical for widespread use in, e.g., low cost consumer audio products.
Another proposed scheme proposes equalizing a sound field by automatically deriving an inverse filter that is embodied in a combination of fast Fourier transforms (FFTs) and finite impulse response (FIR) filters. The inverse filter implementation is quite complex, however, requiring considerable computational resources, thus making this approach impractical also for widespread use in, e.g., consumer audio products. In addition, there is no provision for re-optimization in this scheme.
Therefore, it would be desirable to have a method and/or apparatus for the automatic equalization (i.e., tuning) of the speaker in a hearing aid that does not involve excessive complexity in implementation, such as for example, implementation costs, power consumption and size. It would also be desirable to have a method and/or apparatus for the automatic equalization or tuning of a hearing aid loudspeaker that automatically re-optimizes the equalization. The present invention provides such methods and apparatus.
The present invention provides apparatus and methods for generating digital filters for tuning a hearing aid. Therefore, according to one embodiment, first digital data is provided for a tolerance range for a target response curve of sound level versus frequency for the hearing aid. Second, digital data is provided representing an “audiogram” or a response curve of an individual's sound level perception versus frequency. The first digital data is compared with the second digital data and those responses of the audiogram not within the tolerance range are determined. The parameters for determining a digital audio filter are then generated, and the resulting digital audio filter is applied to the hearing aid to generate a compensated audiogram or response curve. According to other embodiments, iterative audiograms and the determination of one or more additional filters as well as fine tuning of the filter continues until the hearing aid is optimized for the individual. For example, the frequency amplitude and bandwidth of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range, or a predetermined limit on the number of digital audio filters has been reached, whichever occurs first.
These and other features of the invention will be apparent to those skilled in the art from the following detailed description of the invention, taken together with the accompanying drawings.
A typical prior art hearing aid 100 is shown in FIG. 1. The hearing aid 100 is comprised of a microphone 110, audio amplification 120, and a speaker 130. Early hearing aids provided amplification across the audio spectrum. However, there was generally an attempt to pick the center frequency of the amplifier to coincide with the frequency at which the wearer was most likely to be experiencing the most impairment. These early single frequency systems were followed by improved systems which included two, three or even more filters so that specific frequency bands in the hearing aid could be amplified to different levels.
Previously, such systems have been completely analog as is shown in the analog hearing aid 200 of FIG. 2. In
Currently, hearing aids are becoming increasingly digital. A conceptual digital system 300 is shown in FIG. 3. This digital hearing aid 300 is actually comprised of both analog and digital elements since environmental sound inputs and the speaker system 360 is necessarily analog. However, this is referred to as a digital system since the audio signals are all digital prior to reconstruction and playback. The digital hearing aid 300 is comprised of a microphone 310 for receiving sounds occurring in the environment of the individual wearing the hearing aid. The signal output from microphone 310 is provided to an analog to digital converter 320, and then to a digital audio processing function 340, where it is digitally processed. Such digital processing typically includes tuning or selected, amplification of selected frequency bands. Once the digital processing is completed, the signal is converted back to analog by a digital-to-analog converter (“D/A”) 350. The resulting analog signal is then provided to hearing aid speaker 360.
It will also be appreciated by those skilled in the art that although the circuitry of
Of the various hearing aid architectures discussed above, the digital hearing aid of
Due to this flexibility, the current trend in almost every type of audio processing is toward digital functionality. An embodiment of the present invention presents a novel method of tuning, including an automatic method for calculating filter tuning coefficients and applying them in the digital domain.
In a more general sense than discussed above, hearing aid tuning is the process of modifying an audio signal prior to playback by the speaker in order to create a speaker response to compensate for a hearing impairment so that the wearer “hears” sounds as close as possible to those sounds heard by an individual not suffering from a hearing impairment. That is, the speaker output is effectively shaped to compensate for hearing loss by the influence of the tuning process on the audio signal. A flat response of the speaker is normally desired for a public sound system since it allows the sound spectrum to pass through the speaker without audible degradation. However, as discussed above, the speaker output of a hearing aid is almost never flat since hearing loss usually varies with frequency. Although hearing aids are normally tuned to compensate for hearing loss by an individual, it will be appreciated that the sound spectrum of a hearing aid could also be shaped in some way to create a particular sound effect, to enhance intelligibility or the detection of a specific sound, to compensate for a specific listening environment, or provide other audio processing functions as desired.
As discussed above, hearing aid tuning can be performed in either the analog or the digital domain. However, current trends in audio processing and the desirability for smaller and smaller size highlights the need for the inexpensive, highly-flexible, readily-adaptable, digital domain tuning capability.
According to a first embodiment, the audiogram 410 shown in
Once the initial audiogram of an individual is determined, the appropriate filters can be computed and added to a digital hearing aid to compensate or enhance the individual's hearing ability so as to bring the hearing ability as close as possible to a desired ability.
As was mentioned, typically the compensation or enhancement is to improve or correct the hearing ability of an individual having some type of hearing impairment. However, the enhancement could be used to give an individual with normal and acceptable hearing the capability of substantially enhanced hearing at selected frequency bands or in extremely noisy environments.
It should be appreciated that although determining the coefficients for setting one or more filters in the manner as discussed above will provide significantly improved hearing ability to an individual, the hearing aid system can be further tuned by additional fine tuning of the filters determined from the original or initial audiogram, or including additional filters in the hearing aid. To fine tune the hearing aid system, testing of the hearing ability of the individual or generation of another audiogram is accomplished while the individual is wearing the hearing aid with the initial filter settings. This new audiogram of the hearing ability of the individual should indicate significant improvement over the original audiogram. However, the new audiogram may well indicate that the individual's hearing ability still has room for improvement. Thus, the values of the existing filter can be fine tuned and/or new filters added. Still another audiogram can be generated while the individual is wearing the twice tuned hearing aid and the process repeated until no further meaningful improvement is achieved.
According to another embodiment, an air threshold response tuning process may be accomplished by fitting the individual with a completely untuned hearing aid and locating him or her in a sound room such as discussed in FIG. 5. It will also be appreciated by those skilled in the art, that with minor alteration of the equipment shown in
However, also included is a control and data link 675 to digital computer 630 which is, in turn, connected by link 680 to the button or switch 690 in the sound room and used by the individual wearing the hearing aid to indicate he hears the sound at a specific frequency and power level. (See
As discussed, this configuration allows applying the computed filter coefficient to a selected hearing aid by an individual in order to fine tune and verify the effectiveness of the correction. This is desirable since every patient is different and their response to the sounds may not be as expected or predicted even if the hearing aid model is the same and they are all manufactured the same. That is, the patient simply may not respond to an initial correction as predicted. Accordingly, computer controlled system 630 sends a digital audio signal through the hearing aid amplifier and tuning circuit 640 which contains filtering and then to the hearing aid speaker 660. The hearing aid 660 also, of course, includes a D/A converter for converting the resultant filtered digital audio signal to an analog signal suitable for the speaker 660. The speaker output is sound that is sensed by the individual wearing the hearing aid. The individual hears a tone of known frequency and amplitude and sends a signal by means of switch button 690 back to the computer controlled circuitry 630.
The first time through this sequence, the hearing aid filter coefficients are all set as all-pass filters, meaning the filters do not affect the sound level characteristic of the hearing aid. Once the initial measurement has been made, an automatic tuning circuitry 600 in computer controlled circuitry 630 determines a set of filter coefficients which are returned to the hearing aid 600 being fitted to the individual to tune the hearing aid output. Methods for determining these coefficients are discussed below.
The coefficients thus determined are transmitted to filters in the hearing aid and are used to change the characteristics of the filters which are applied to the received digital audio signal. The evaluation cycle is performed again with these new coefficients in place. The new measurement shows the improvement made by the filters thus calculated. If the new measurement shows the need for further correction, the computer controlled circuit 630 computes additional coefficients and the cycle is repeated.
The equalization coefficients for the system of
Preferably, a second order infinite impulse response (“IIR”) equalization filter algorithm is used as the digital filter algorithm for the algorithm shown in FIG. 7. On the other hand, the particular method for determining the coefficients of such second order IIR equalization filters is not of critical importance to the invention. Any digital filter algorithm for generating filters, FIR or higher order IIR filters, determined by the coefficients of the equations specifying the filters, may be used in this method. In addition, the method for estimating A, Fc and BW is not critical, since the single filter optimizer, described below, reiteratively adjusts these values. Furthermore, it should be understood that FIR filter or other techniques may be employed, if so desired.
In the preferred embodiment the value of Fc is initialized to the frequency where maximum deviation from the desired response curve occurs, e.g., as found in step 720, below, and A is initialized to the negative of that maximum deviation. BW is estimated by determining the 3 dB dropoff point from the maximum deviation, and using that value.
For example, suitable methods for determining the filter coefficients are disclosed in Orfanidis, S J, “Digital Parametric Equalizer Design with Prescribed Nyquist-Frequency Gain,” 101st Audio Engineering Society Convention, 4361 (I-6), Nov. 8-11, 1996, Los Angeles, Calif., and in Bristow-Johnson, R., “The Equivalence of Various Methods of Computing Biquad Coefficients for Audio Parametric Equalizers,” AES, 97th Convention, 3966 (K-6), November 1994, are both applicable. The method of Zolzer and Boltze (99th Convention of AES, Oct. 6-9, 1995) (hereinafter referred to as “the Zolzer and Boltze article”) is also applicable, and is perhaps more readily comprehended. It provides a method for generating coefficients for a five coefficient bi-quadratic discrete-time filter. Key aspects of the method presented in the Zolzer and Boltze article are summarized here, for exemplary purposes:
Let:
Now, if a filter with a positive dB gain, i.e., having a linear gain of greater than one (known as a ‘boost’ filter), is to be designed, aB is computed according to Equation (54) of the Zolzer and Boltze article, which can be rewritten by substituting from the above to be:
If a filter with a negative dB or fractional linear gain (known as a ‘cut’ filter) is to be used, the value ac is computed according to Equation (55) of the Zolzer and Boltze article, which can be rewritten by substituting from above to be:
Equation (56) of the Zolzer and Boltze article is rewritten as:
d=−cos(2πFc/Fs)
H0 is computed according to Equation (58) of the Zolzer and Boltze article:
H0=V0−1
Referring to
Where aBC is aB for the boost case, or aC for the cut case. Those skilled in the art will recognize that this equation is a complex function and thus comprehends both phase and magnitude. Although only the speaker's magnitude response of the hearing aid is being considered, the filters themselves are dealt with as complex functions, and so phase and magnitude interactions are dealt with in the derivation of the compensating filter scheme. Those skilled in the art will also recognize that the filter coefficients then have the following forms:
Included below is a Matlab function for computing the coefficients in the manner indicated:
function [B, A]=apcoef(A, BW, Fc, Fs)
if A<1
a=(tan(pi*BW/Fs)−A)/(tan(pi*BW/Fs)+A);
else
a=(tan(pi*BW/Fs)−1)/(tan(pi*BW/Fs)+1);
end
H=A−1;
d=−cos(2*pi*Fc/Fs);
b0=1+(1+a)*H/2;
b1=d*(1−a);
a1=b1;
b2=(−a−(1+a)*H/2);
a2=−a;
B=[b0 b1 b2];
A=[1 a1 a2];
As an example, let A=2, Fc=1000, BW=500, and Fs=44100:
B=[b0 b1 b2]=[1.03440794155482−1.91161634125903 0.89677617533555]
A=[1 a1 a2]=[1.00000000000000−1.91161634125903 0.93118411689037]
The automatic hearing aid algorithm of
Step 710:
The single filter optimizer operation referred to in step 750 of
Step 1010:
The joint filter optimizer operation referred to in step 770 of
Step 1110:
Set i=0 to begin.
Step 1120:
Increment i. If i becomes greater than the number of filters, n, make i=1.
Step 1130:
If not, stop. Optimization is complete.
Although the present invention and its advantages have been described in detail, it should be understood that various changes, substitutions and alterations can be made herein without departing from the spirit and scope of the invention as defined by the appended claims. For example, while the joint filter optimization subroutine 720 discussed above in conjunction with
This invention is related to Ser. No. 09/191,944 filed Nov. 13, 1998, and having common inventors with the present application. application Ser. No. 09/191,944 is incorporated in its entirety into this application.
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| Number | Date | Country | |
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| 20030063763 A1 | Apr 2003 | US |
