Hearing aid with digital processing for: correlation of signals from plural microphones, dynamic range control, or filtering using an erasable memory

In an illustrated embodiment a behind-the-ear hearing aid includes a microphone, an amplifier-low pass filter circuit, an analog to digital converter, a digital integrated circuit arithmetic and logic unit for implementing a n-th order transfer function in the Z domain, a digital to analog converter and an output transducer, for producing the desired sound response. A memory multiplexer is provided for loading of the multiplier coefficients necessary to adapt the transfer function circuit to essentially any class of hearing deficiency into an erasable programmable read only memory (EPROM). The structure is such that the coefficient memory may be loaded after the standard universal hearing aid has been completely assembled, and indeed the hearing aid may be reprogrammed as needed after a period of use, essentially without disassembly.

BACKGROUND OF THE INVENTION 
The invention relates to a method for adapting the transmission function of 
a hearing aid to various types of hearing difficulty, and to hearing aids 
for the implementation of this method. A device of this general type is 
known from the German Patent No. 15 12 720. 
With conventional hearing aids, there are problems in being able to adapt 
the characteristic data as well as possible to the individual hearing 
impairments of a person with difficulty in hearing. The electrical 
properties of hearing aid amplifiers are determined by the structural 
elements used in the construction and at most can only be varied to a 
slight extent by external controls. This means that there must be a 
plurality of hearing aids which differ from one another for instance only 
in the frequency response to the amplifier. 
Hitherto, therefore, it has not been possible to find a uniform form of 
construction for hearing aids. At the present time alone there are several 
hundred models on the hearing aid market which can be sorted into classes 
only by consideration of individual parameters. 
A further series of types must be adapted to the dynamic range of an 
afflicted hearing, this range being changed, for example restricted, with 
various types of hearing difficulty. These hearing aid amplifiers have 
additional control loops in order to be able to adjust the output level of 
the hearing aid to the limits suitable for the hearing for which provision 
is to be made. 
According to one particular construction, such as is described for example 
in the German Offenlegungsschrift No. 23 16 939, an adaptation can also 
be effected by the frequency range transmitted by the hearing aid being 
split into at least two partial ranges, to each of which there is 
coordinated a separate level control acting independently of the other 
frequency ranges, with one or more control loops in each case. This 
construction also results in an extensive system of structural elements, 
so that there are difficulties in obtaining the small construction which 
is both customary and desirable in hearing aids. 
SUMMARY OF THE INVENTION 
The invention proceeds from the assumption that the transmission function 
of a hearing aid is essentially determined by the properties of the 
transducers, the amplifier electronics and the physical dimensions of the 
sound inlets. They are determinative: (a) for the frequency response; (b) 
for the input-output dynamics; and (c) for the transient response. 
Re (a): 
The frequency response of a hearing aid is prescribed by the choice of the 
structural elements in a conventional hearing aid amplifier. If this 
frequency response is to be controlled by adjusting controls, the 
possibilities for so doing in the hearing aid are very restricted by the 
confined space conditions. The confined space virtually allows only a 
simple tone control or sound balance. The effectiveness of these adjusting 
controls is limited, since filter slopes greater than 12 dB/octave are not 
possible due to the known lack of space. 
Re (b): 
The input-output dynamics of a hearing aid should be able to be adapted as 
well as possible to the dynamic behavior of the hearing which is to be 
amplified. For this purpose the known PC (Peak-Clipping) limiting circuits 
and AGC (Automatic Gain Control) control circuits are used; the first are 
static adjusting controls, whilst the second possibility is a dynamic 
control. This brings us to the third point. 
Re (c): 
Each control is time-dependent; automatic adjustment of the amplification 
is not effected inertialessly. 
The aforementioned points show that a "standard hearing aid amplifier" must 
therefore display all the aforesaid properties. With the present 
structural elements, the number of adjusting controls and control elements 
would be such that it would be impossible to manufacture a device to be 
worn on the head, for example behind the ear. Using amplifiers of known 
construction and corresponding design the space requirement cannot be met 
in these devices. 
With a method for adapting the transmission function of a hearing aid to 
various types of hearing difficulties, it is an object of the invention to 
disclose a simple construction which can be accommodated in small devices 
and which is at the same time very effective as regards hearing defects to 
be compensated. According to the invention this object is solved by a 
process characterized in that the analogue sound signal to be transmitted 
is converted into a digital signal, is then subjected to a discrete signal 
processing based on selected stored parameters matched to the difficulty 
in hearing for which provision is to be made, that the digital signal is 
then converted back into an analogue electrical signal and is converted 
into sound in a manner known in the case of hearing aids. 
An adaptation to the requirements of a hearing aid for the hard of hearing 
can be obtained in simple manner through the principle in accordance with 
the invention, i.e. the adjustment or control, i.e. alteration, of the 
transmission function of hearing aids effected by an arithmetic unit. This 
construction permits the parameters determining the frequency response and 
the dynamic behavior to be stored in suitable memory locations in the form 
of numerical values. In contrast to known electronic amplifier hearing 
aids, the new devices can be regarded as digital or computer hearing aids. 
With these, there is also achieved the advantage that parameters 
determining the transmission function of a hearing aid which have been 
read into a memory can also be modified again, i.e. one is not bound to a 
specific amplifier structure. The invention introduces a standard hearing 
aid wherein all the necessary transmission functions can be adjusted on 
the finished device after assembly has been completed. 
A memory to be used may in this instance be designed such that it is 
charged only when the hearing aid is adapted to the afflicted hearing. 
This may be a single occurrence or, when using suitable erasable memories, 
can be altered as required. In American usage such memories are called 
"erasable programmable read only memory" and, in abbreviated form, 
"EPROM". An extensive variability of adaptation of hearing aids is 
particularly important for subsequent corrections of characteristic 
curves. 
A memory which can be used in accordance with the invention should, for 
example, have the form of known microprocessors, of which one is described 
e.g. in the pamphlet "DAC-76" of the firm Precision Monolithics Inc., 1500 
Space Park Drive, Santa Clara, California 95050. With this construction, a 
memory can also be built into a hearing aid worn on the body and operated 
there. The transmission behavior of a hearing aid, which results from the 
properties of the transducers, i.e. microphone and earphone, and that of 
the amplifier; i.e. the transmission function of the device 
(characteristic curve), the amplitude of response to each input frequency 
component, which appears again e.g. as a received frequency at the hearing 
aid output, and/or the ratio of the input level to the output level, is 
controlled according to the invention by means of an arithmetic unit such 
that the input signals are altered for the purposes of compensation of a 
hearing defect; for example, adaptation to a sensitivity of hearing which 
is changed relative to occurring frequencies, for example, a narrower pass 
band, and adaptation to changed dynamics. The arithmetic unit should 
therefore additionally have a memory. An upper limit to the number of 
memory locations is given by the required upper cutoff frequency of the 
transmitted low frequency band. According to the invention, it is possible 
to alter all incoming sound signals in the desired manner such that the 
changed transmission function desired is achieved. 
Signals which can be processed are obtained in the manner customary with 
hearing aids, in that the signal coming from the microphone is supplied to 
an amplifier and a low pass filter. The signal thus preliminarily treated 
is then supplied to an analogue-digital converter and converted into 
signals which can be processed with a computer transmission function H(z) 
in an arithmetic unit. This unit can contain, stored, the parameters which 
are to determine the transmission behavior of the system. A signal is then 
obtained from the arithmetic unit which, supplied to a digital-analogue 
converter for suitable conversion to analogue form, and, if necessary, 
after passing through a terminal amplifier, and supplied to an output 
transducer, for example an inserted earphone, is suitable for supplying 
sound which is adapted to the afflicted hearing. 
Adjustment of the transmission function of the arithmetic unit can take 
place, for example, by way of a memory multiplexer. This is, as known, a 
structural element with which it is possible to selectively or 
sequentially load several memory locations by way of only one line. The 
incoming signals themselves can be used for effecting sequential address 
control. Establishing the parameters can be effected in the conventional 
manner by way of an audiometer. In an ideal development, the measured 
values determined in an audiometer can be transmitted directly via a 
memory multiplexer into the memory of the arithmetic unit for storage in 
the memory. 
Further details and advantages of the invention will be explained 
hereinafter with reference to the exemplified embodiments illustrated in 
the accompanying sheets of drawings; and still further objects, features 
and advantages will be apparent from this detailed disclosure and from the 
appended claims.

DETAILED DESCRIPTION 
FIG. 1 shows a block circuit diagram of a hearing aid with discrete signal 
processing. It comprises as input sound converter a microphone 1 of known 
construction which is supplemented by an amplifier 2. Using known TTL 
elements, energy sources with five volt (5 V) supply voltage can be used 
and with CMOS elements the voltage can be dropped to 1.5 V. The energy 
requirement therefore varies within a scope which can be satisfied even in 
the case of hearing aids. 
The amplifiers 2 to be used in accordance with the invention operate at the 
same time as low pass filters 3 in order to present a limited signal to 
the following analogue-digital converter 4. The upper cutoff frequency of 
this signal should be less than half the sampling frequency. The known 
Sampling Theorem states that the sampling frequency should be fixed at 
least twice as great as the highest occurring signal frequency. If this is 
disregarded, the effect known as aliasing occurs, i.e. higher frequency 
components are reflected about the angular frequency. Depending on the 
type of analog-digital converter used, a holding circuit, not separately 
illustrated, is required before the conversion, the latter circuit holding 
the signal stable for the time required for the conversion. 
A further block 5 identified with H(z) is connected to the analog-digital 
converter 4. In this block 5, the signal which occurs as input signal 
U(z), is controlled such that the output signal Y(z) is the product of 
U(z).times.H(z). 
In this instance, U(z) can be directly the numerical sequence generated at 
the output of the analog-digital converter 4. It may, however, 
particularly if a volume control is intended, be a modified numerical 
sequence which results in a correspondingly modified limited input-output 
characteristic curve. One possible method of obtaining the input-output 
characteristic curve would be to multiply the input value with the 
characteristic curve value; another method, particularly rapid in digital 
technology, would be to pick up the number produced by the analog-digital 
converter 4 as an address for a memory. The output value then lies in the 
memory location indicated by the address. This method is particularly fast 
and, with eight bit words, only requires 256 memory locations. Such a 
memory may be taken as included within the component 4 or 6 of FIG. 1. 
For realization of the function, the block 5 contains memories, multipliers 
and adders. If care is taken that the computing time of the multipliers is 
fast enough, all the multiplications can run over one multiplier with the 
use of time division multiplexing. There need not then be a multiplier for 
each multiplication. 
If an upper signal band width of 6 kHz is judged satisfactory, a sampling 
frequency of at least 12 kHz results. With a factor of 2.3 there results a 
sampling frequency of 13.8 kHz or a time of 72.5 .mu.sec between two 
values of the numerical sequence U(z). For the multiplication and addition 
of two eight-bit numbers, times of 115 nanoseconds are possible. This 
means that a single multiplier and adder can effect 630 operations in the 
time between two sampling values. This means that, with this construction, 
the transmission function can have up to 630 poles and zero positions. 
To the output Y(z) of the transmission function H(z), i.e. the block 5, 
there is connected a digital-analog converter 6 which converts the 
discrete signal into a continuous signal. This signal is supplied to a 
receiver 8 via a terminal amplifier 7. 
The parameters determining the transmission behavior of the device do not 
have to be fixed at the time of manufacture of the device. They can be 
determined at the actual time of adapting the device to an ear with 
impaired hearing, i.e. at the moment at which the charging of the memories 
(e.g. the loading of parameter values into EPROM 13-19, FIG. 2) also 
actually needs to be carried out. A memory multiplexer connected via a 
line 11 (FIG. 2) which is drawn in the block circuit diagram and 
designated by 12 (FIG. 2) can generally serve this purpose. This memory 
multiplexer 12 allows the parameter values to be read into the block 5 
serially. These parameter values can be optimally fixed on the basis of 
audio-metrically determined characteristic data of the hearing for which 
provision is to be made. 
In FIG. 2, to clarify its function, the block 5 of the memory computer unit 
is enlarged and emphasized with details. In this instance, the two 
connections to the converters 4 and 6 of FIG. 1 are indicated by the 
connecting points 9 and 10. The block 5 has a further connection 11 
through which the parameters of the desired transmission function are 
introduced. A particularly accurate adaptation can be effected in that the 
audiogram is put into a form which is readable for the block 5 and this is 
then read into the block 5 by way of a multiplexer 12 in a manner known in 
computers. The multiplexer 12 controls the memory points in desired 
sequence, i.e. in the present case, the memory point 13, etc. to 16 first. 
Subsequent to this, reading into the points 17, etc. to 19 likewise 
follows. This reading-in of the parameters a.sub.o to a.sub.n and b.sub.l 
to b.sub.m is indicated by the arrows 20 to 26. The letters n and m in 
these expressions may each stand for the number four, respectively, 
corresponding to four parameters, according to which, in the present case, 
an adequate processing of the input signal can be effected. Further, the 
block 5 also contains discrete signal processing components 27 to 32. 
Function points in which the signals coming from 9 or 27 to 32 are 
processed corresponding to the parameters from storage locations 13 to 19 
are indicated by circles 33 to 41. An output signal Y(z) can then appear 
at 10 by way of the coupling points illustrated as circles 40 and 41; the 
output signal, as indicated above, is altered by calculation in a known 
manner corresponding to the stored parameters such as a.sub.o to a.sub.n 
and b.sub.l to b.sub.m. This signal can then be treated in the manner 
customary with hearing aids, specified in FIG. 1, and can be supplied to 
the ear. 
The memory, i.e. the points or storage locations such as 13 to 19, can be 
constructed such that it can be erased by ultraviolet (UV) light or by 
electrical means. The invention thus offers a universally applicable unit 
for the manufacture of hearing aids. 
As a result of the new method of signal conversion in the hearing aid, i.e. 
as a result of the discrete signal processing, it becomes possible to 
design the transmission function H(z) such that several input signals, for 
example those of two pick-up microphones, can be processed. In this way, 
the (two) inputs can be correlated with one another and an output signal 
obtained which has a substantially higher signal to noise ratio than is 
possible with only a single signal path. 
The input of plural analog sound signals for individual conversion to 
digital signals and for correlation in a digital arithmetic unit to 
provide a resultant digital output is indicated in FIG. 7 by means of 
component 50. Thus the plural inputs to the component 50 are correlated 
with one another and an output signal obtained which has a substantially 
higher signal to noise ratio than is possible with only a single signal 
path. 
By way of example, component 5 of FIGS. 1 and 2 may be implemented as an 
integrated circuit microprocessor. Where the hearing aid receives several 
input signals e.g. from a microphone (such as shown at 62, FIG. 3), 
pick-up induction coils (such as shown at 62a in FIG. 3), etc., the input 
signals are individually converted to digital signals (as indicated by the 
legend applied to input means 51 in FIG. 7), whereupon the discrete signal 
values corresponding to essentially the same instant of time are 
correlated in the microprocessor to improve the fidelity of the resultant 
digital input signal based thereon, which resultant digital signal is then 
subjected to the processing step of component 5, FIGS. 1 and 2. By this 
means (as represented in FIG. 7), an improved signal to noise ratio may be 
achieved. Thus the component 50 in FIG. 7 may be described as analog to 
digital conversion and discrete signal correlation means. The component 5 
may be designated time domain discrete signal processing means as 
indicated by the label applied to this component in FIG. 7. 
The memory used for components such as 13-19, FIG. 2, may be an operational 
part of the microprocessor and may be an erasable, electronically 
programmable read only memory, which can be electronically loaded with the 
selected parameters after it has been fully packaged as a behind-the-ear 
hearing aid, via a conventional memory multiplexer as indicated at 12 in 
FIG. 2. Preferably, the hearing aid may be reprogrammed after a period of 
use, as necessary, without any substantial disassembly of the hearing aid. 
Thus, for example, terminals such as 11, FIG. 2, and any other portion of 
the memory necessary to the erasure and reprogramming operations may be 
readily accessible from the exterior of the hearing aid. 
An audiometer is shown in German Patent No. 10 16 894, and an improved 
audiometer operable by means of coded signals from a microprocessor is 
shown in my U.S. application for patent Ser. No. 888,843 filed Mar. 22, 
1978, and corresponding to German Application No. P 27 19 796.2 filed May 
3, 1977. 
Circuit Description of a hearing aid following the greater detailed 
illustration of FIGS. 3 to 6 whereby: 
in FIG. 3 is shown the input stage of this hearing aid and A/D-converter, 
in FIG. 4 is shown EPROM dynamic range compression, 
in FIG. 5 is shown timemultiplexing of input values (for a FIR-filter 
length, 53 i.e.h (.nu.) for .nu.=0(1),32) and 
in FIG. 6 is shown a time multiplexed multiplier and accumulator, 
D/A-converter. 
In the input stage of the hearing aid an input transducer like a microphone 
62 or a telephone coil 62a gives a signal which is amplified and band 
limited in the two transistors 63, 64 amplifier stage. The continuous 
analog signal is sampled and held in a sample and hold amplifier 65 (Burr 
Brown SHC 80 KP or equivalent). 
The sample impulse is taken from the END OF CONVERSION impulse of the A/D 
converter. The A/D (analog to digital) converter is built with a 
comparator 66 (CMP 01 from PMI) two exclusive OR-gates 67, 68 (2.times.1/4 
7486) one D-flip flop 69 (1/2 of 7474) and one successive approximation 
register 70 (AM 2502), and one digital to analog converter 71 (COMDAC -76 
from PMI). Each analog signal conversion results in an 8-bit digital word. 
The 8-bit word of the A/D converter of FIG. 3 (output "8 data bits") is 
loaded into the input of 8-bit latch 72 (74100) of FIG. 4 with the END OF 
CONVERSION signal from the successive approximation register. The output 
lines of this latch are connected with the address lines of an erasable 
and programable read only memory 73 EPROM (2708). 
The contents of this memory translate the 8-bit data word from the A/D 
converter into a 12-bit data word as used in further computations. The 
relationship between the input and output data word is such that all 
dynamic compression needed to fit a particular hearing damage, is stored 
as a table in the EPROM memory 74. 
One implementation of the transfer function H(z) could be a finite impulse 
response-(FIR-) filter of FIG. 5. This FIR-filter can be implemented using 
only one multiplier in a time multiplexed configuration. This is called 
time multiplexing of the input signal in the literature. The 12-bit input 
signal is connected to the A-inputs of a 2:1 multiplexers 75 to 77 
(74LS157). The output of that multiplexers 75 to 77 are connected to shift 
registers 78 to 125. 
Shown are 4 times 8=32 stages in each of the 12 rows 78 to 81, 82 to 85 and 
so on of shift registers 78 to 125. This is sufficient for a FIR-filter of 
degree 32. All outputs 126 to 137 have connections with multiplexer 75 to 
77 B inputs 139 to 150. In FIG. 5 only the connection of the first output 
126 to B input 139 is shown and indicated as 152. The multiplexers 75 to 
77 inputs B are active during 31 of the 32 shift pulses. At the 32th pulse 
inputs B are deactivated and inputs A coming from the output "Data for 
FIR-filter" of FIG. 4 and entering multiplexers 75 to 77 at the lines with 
the same numerals (nos. 1-12 respectively) as indicated at the output of 
FIG. 4, are activated. This shifts a new data word into the shift 
registers 78 to 125. At the same time the oldest data word, that is 32 
sample pulses old, is lost. It is no longer needed in the computational 
process. 
The outputs at 153 to 164 of the shift registers 78 to 125 and connected to 
153' to 164' of the inputs of the hearing aid parts (input from time 
multiplexing SR) are the 12-bit input word for the multiplier X-input port 
166 (FIG. 6). The output of an EPROM (2708) 167 is the 12-bit input word 
for the Y-input 168 of the multiplier 165. The contents of this EPROM 
memory 167 (Filter coefficients memory) control the transfer function 
H(z). At 169 all signals multiplied at 165 of one row 78 to 81 etc. are 
added. Every time the multiplexer gates A are activated, the contents of 
the multiplier accumulator 170 are latched into an output latch 171 and 
the accumulator is cleared. The output of that latch is the input for a 
digital to analog (D/A) converter (AD7521) 172. The output of that D/A 
converter is low pass filtered and amplified in the final stage of the 
hearing aid amplifier. This final stage drives the hearing aid output 
amplifier 7 and transducer 8 of FIG. 1. 
It will be apparent that many modifications and variations may be effected 
without departing from the scope of the novel concepts and teachings of 
the present invention.