Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
  • H04R25/00—Electric hearing aids
  • H04R25/45—Prevention of acoustic reaction, i.e. acoustic oscillatory feedback
  • H04R25/453—Prevention of acoustic reaction, i.e. acoustic oscillatory feedback electronically
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
  • H04R25/00—Electric hearing aids
  • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
  • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing

Definitions

• the invention concerns a digital hearing aid as diclosed in more detail in the preamble to claim 1.
• a hearing aid of this kind with digital suppression of or compensation for acoustic feedback is known from the ap- plicant's earlier European patent application no. 90309342.5 (publication no. EP-A2-0415677).
• the present ap ⁇ plication is related to this European application, which was filed on August 24th 1990, and everything disclosed in said patent application therefore forms part of the present application with this reference.
• Such a hearing aid has in practice proved to function as intended.
• the compensation which is carried out by updating the co ⁇ efficients in a digital filter in a feedback circuit, is effected by means of an algorithm which takes into account the error in the filter, i.e. the difference between the filter's actual setting and the desired setting.
• Such a hearing aid will not always be quick enough to adapt to sudden changes in the acoustic feedback path, even though it is still able to compensate for the acoustic feedback which arises.
• the lack of speed in the adaptation function can result in undesired acoustic signals which can be heard by the user of the hearing aid.
• the object of the present invention is to increase the adaptation speed without hereby giving rise to any incon- venience for the user of the hearing aid.
• the algorithm which takes care of the updating of the* coefficients in the digital filter in the compensation cir ⁇ cuit must take into consideration that the filter error depends on the number of coefficients, signal/noise ratio, input level, volume, and on the degree of peak clipping in the limiter circuit.
• Such an embracing algorithm will not be particularly fast in adapting itself to changes in the acoustic feedback path, but on the other hand it will pro- vide a reliable and precise adjustment of the filter under stationary conditions in the feedback path.
• the updating of the coeffi- cients in the filter takes place following a choice between a number of algorithms, so that the hearing aid selects between alternative algorithms to the basic algorithm when a significant change occurs in the acoustic feedback path. If, for example, the change is such that a greater acoustic feedback occurs, the hearing aid immediately selects an algorithm with a greater speed of adaptation. This happens, for example, by adding more noise and/or increasing the adaptation speed in excess of what is prescribed by the basic algorithm. The quick condition lasts until the cir- cuit ascertains that the filter coefficients are stable again, after which the circuit automatically switches back to the basic algorithm for continuous adjustment of the electronic compensation.
• a hearing aid with such a coupling between a general algo ⁇ rithm and a quick or quicker algorithm is able to react considerably more quickly to a significant change in the acoustic feedback path, as compared with an aid which func ⁇ tions solely with the general algorithm, even though with the hearing aid according to the invention there is intro ⁇ quizd, for example, 6 dB less noise in the general algo ⁇ rithm.
• the measurement of changes which are so great that the cir ⁇ cuit shifts from the general algorithm to an alternative, e.g. a quicker algorithm, is effected preferably by a stat ⁇ istical monitoring of the coefficients in the digital fil- ter. For example, a significant change occurs in the acous ⁇ tic feedback path when one or more coefficients during the change comes out in excess of 4 x the calculated standard deviation.
• fig. 1 shows a block diagram of a hearing aid according to the invention
• fig. 2 shows a more detailed presentation of the block diagram in fig. 1,
• fig. 3 shows a block diagram of the adaptation part of the hearing aid in figure 1 and 2
• figs. 4 and 5 show block diagrams of pseudo-random-binary generators and a variant hereof, and
• fig. 6 shows a block diagram of the noise level control circuit in the hearing aid in figure 2.
• a hearing aid comprising a sound re ⁇ ceiver, for example in the form of a microphone 5, a pre- amplifier 7, a digital adaptation circuit 3, an output amp ⁇ lifier 9 and a sound reproducer 11, for example a miniature electro-acoustic transducer.
• the preamplifier 7 is of a commonly-known type, for example of the type known from the applicant's earlier European application no. 90309342.5, and the output amplifier is similarly of a commonly-known type, for example correspond ⁇ ing to the output amplifier which is used in the hearing aid in the applicant's earlier European application no. 90309342.5.
• the digital adaptation circuit 3 is shown within the stip ⁇ pled frame in the connection 13 between the preamplifier 17 and the output amplifier 9. However, there is nothing to prevent the circuit 3 from being a mixed analogue and di ⁇ gital circuit, but in the preferred embodiment a purely di- gital circuit is used.
• the input to the digital adaptive circuit 3 comprises an A/D converter 17 and the output from the circuit comprises a D/A converter 19.
• a digital limiter circuit 15 of a known kind, for example as known from the" applicant's earlier Europen application no. 90309342.5.
• the function of the limiter circuit 15 is to prevent the electrical signal from reaching a level of amplitude which exceeds the linearity limits of the output amplifier 9 and the transducer 11, and as explained in said European ap ⁇ plication.
• a digital summing circuit 21 is inserted in the path be ⁇ tween the limiter circuit 15 and the D/A converter 19.
• the summing circuit 21 serves as a place for the introduction of a noise signal N as explained later.
• a digital subtrac ⁇ tion circuit 23 is inserted in the path between the A/D converter 17 and the limiter circuit 15.
• the subtraction circuit 23 comprises means for the introduction of elec ⁇ trical feedback, as will also be described later.
• the normal signal path for a desired signal from the micro- phone 5 to the transducer 11 is the direct circuit path a- b-c-d-e-f-g-h as shown in fig. 1.
• the electrical path a, b, g and h is arranged for analogue signals and thus normally comprises only a single con ⁇ ductor, while the electrical signal path c, d, e and f is arranged for digital signals and will thus comprise a num ⁇ ber of parallel conductors, for example 8 or 12 conductors, depending on the bit number from the A/D converter 17.
• Electrical feedback is derived from a tap 25 in the area f in the digital signal path between the summing circuit 21 and the D/A converter 19, which means that the electrical, digital feedback signal comprises a noise-level component.
• the feedback signal is led through an adaptive filter 27 which is shown as a "limited impulse response filter", a so-called FIR filter (Finite - Impulse - Response filter), and after passing through this filter, the feedback signal is fed to .the digital subtraction circuit 23 via a digital signal path m.
• the digital signal from the tap " 25 is fed via a delay circuit 29 before being fed to the FIR filter 27 as a digital signal 41 via the digital lead k.
• the delay in the delay circuit 29 is of the same order as the minimum acoustic path length between the transducer 11 and the microphone 5, and must introduce a delay which corresponds hereto. It is not necessary to introduce such a delay by means of the delay circuit 29, but significant re- dundancy in filters and correlation circuit is hereby avoided, so that the overall circuit is simplified.
• the im ⁇ pulse response from the filter 27 is continuously adjusted, controlled by coefficients from a correlation circuit 31.
• the correlation circuit 31 constantly seeks for correlation between the inserted digital noise and any noise component in the residual signal in the connection d after the di ⁇ gital subtraction circuit 23.
• the inserted noise signal N is generated from a noise source 33 and is introduced via the digital summing circuit 21 after level adjustment in the regulation circuit 35.
• the noise signal is also coupled to a reference input on the correlation circuit 31 via a second delay circuit 37, which also introduces a de ⁇ lay of the same order as the minimum acoustic path length between the transducer 11 and the microphone 5 via a signal path n.
• the residual signal on the lead d constitutes the input signal on the correlation circuit 31, in that the signal is fed hereto from a point 39 on the lead d and by means of the digital lead 57.
• a circuit 79 in the form of an algorithm control circuit which determines the algorithm in accordance with which the correlation cir ⁇ cuit 31 must send coefficients further to the filter 27, in that the algorithm control circuit 79, via the digital con ⁇ nections 80, 81, constantly monitors and controls the cor- relation circuit 31.
• the algorithm control circuit 79 also controls the supply of digital noise from the noise gener ⁇ ator 33 by regulating the level in the circuit 35 via the * lead 82.
• the residual signal is fetched from the tap 39 via the lead 84, the amplitude of the noise signal is fetched via the lead 83, see figure 2, and the volume signal is fetched via the lead 86, which is explained later.
• the electrical output signal from point 25 is thus fed via the delay circuit 29 to the adaptive filter 27 (FIR), and to the subtraction circuit 23 as the final feedback signal, where the subtraction from the input signal is carried out.
• the feedback signal will corres ⁇ pond completely to an undesired acoustic feedback signal which, via a feedback path w, is conducted from the trans ⁇ ducer 11 to the microphone 5. If the feedback signal and the signal from the acoustic feedback are completely ident ⁇ ical, there will be no residual signal from the acoustic feedback on the lead d, the reason being that the digital feedback signal from the lead m will completely cancel out the acoustic feedback signal.
• the noise signal N is added to the output signal via the summing circuit 21 after level regulation in the circuit 35.
• the noise signal will thus exist in both the inner feedback circuit 3 and the outer acoustic feedback path w.
• the noise signal will thus pass the D/A converter 19 and, via the amplifier 9, reach the transducer 11 and be con- verted to an acoustic signal which is superimposed on the desired signal.
• the level of the noise signal is set in such a manner that it is of no inconvenience to the user of the hearing aid.
• the two said signals do not cancel each other out completely, and a small amount of noise and other feed ⁇ back signals are to be found in the residual signal on the digital lead d, and these are detected by the correlation " circuit 31 which constantly looks for correlation between the residual signal and the delayed version of the noise signal n.
• the output signal from the correlation circuit 31 is an expression for the residual signal, and is used for controlling the filter 27 by changing the filter coeffi ⁇ cients.
• the adaptation is thus arranged that the filter 27 is constantly adjusted so that the feedback system seeks towards a situation in which the noise is cancelled.
• Phys ⁇ ical changes in the environment for the hearing aid and its user, and limitations in the algorithm which controls the system give rise to the result that complete cancellation cannot always be achieved, which is why the algorithm con- trol circuit 79 is inserted.
• noise signal N with a certain spectral characteristic, i.e. with constant level over the whole of the frequency range over which the hearing aid is arranged to operate, a so-called white noise signal.
• pseudo-random-binary-sequence noise signals with suitable bit repetition can be used. These noise signals can easily be generated, for example by using the circuit shown in figure 4, i.e. by using a clocked shift register 103 with multiple feedback via an exclusive OR-gate 105. Such a cir ⁇ cuit will generate signals with a pattern which is repeated after every 2 M-1 bits, where M is the number of stages in the generator.
• Satisfactory noise signals axe achieved with a repetition length from 127 samples to 32,767 samples, i.e. by using circuits with 7 to 15 stages.
• the choice of noise signal is based on the desire to have a low auto-correlation over any span of time which is of the same magnitude as the time constant of the adaptation cir- cuit, i.e. typically about one second. If the acoustic feedback signal is periodic, for example a sine-wave sig ⁇ nal, stable cancellation is not always achieved, in that in ' such situations the adaptation circuit can wander, which can result in signals which can be heard by the user. Such effects can be eliminated by an increased randomisation in the noise generator. This is shown in fig.
• a hearing aid according to the invention shown in fig. 2 of the drawing, and comprising a user- operated volume control 73 and a similarly user-operated adjustment rheostat 75 for the setting of the level in the limiter circuit 15.
• a volume control which can be operated by the user. This can be placed in the microphone amplifier or in front of the output amplifier, but in both cases the adaptive filter 27 must change its coefficients when the setting of the volume control is changed.
• a multiplication amplifier 77 between the tap 39 and the amplitude limiting circuit 15.
• the amplifier 77 is coupled to the volume control 73 via an A/D converter 67, and from the input to the amplifier 77 there is a digital lead 86 for the algorithm control cir ⁇ cuit 79 so that this circuit can scan the volume setting.
• the amplitude limiter circuit 15 can also be user-operated, in that the potentiometer 75 is coupled to the amplifier 15 via an A/D converter 69. It is desirable that the limiter 15 is user-operated, since the limiting circuit determines the maximum sound-pressure level which can be applied to the user's ear. The output level can be reduced without re ⁇ ducing the gain of the amplifier, which is of significance. The maximum positive and negative sound pressure is thus regulated by the user with the potentiometer 75.
• Fig. 2 also shows that the two potentiometers 73 and 75 are con ⁇ nected to a common source of reference voltage 71.
• the level of the inserted noise can be regulated to obtain optimum adaptation.
• the amplifier 35 after the noise generator 33 is controlled by a computation unit 65, for example in the form of a single-stage recursive filter, for example shown in fig. 6.
• the unit 65 is coupled via the two-way connec ⁇ tion 82, 83 to the algorithm control unit 79, so that the unit 79 can fetch the noise amplitude from the unit 65, and such that the signal/noise ratio can be regulated by the algorithm control unit 79.
• the input to the unit 65 is taken from point 63 (see fig. 2) in the connection between point 39 at the input to the correlation circuit and the noise insertion circuit 21.
• the computation unit 65 has a multi- value output signal which is a function of the level at point 63, and is selected in such a manner that the sum of the desired signal from the limiter circuit 15 and the noise signal added hereto does not exceed the saturation level in any of the components which follow after, espe ⁇ cially the summing circuit 21, the D/A converter 19, the output amplifier 9 and the transducer 11.
• the recursive filter 65 is of the first order and compris ⁇ es, as shown in fig. 6, a first circuit 111 for the meas ⁇ urement of the absolute signal level. This is followed by a* first multiplier 113 which produces an output signal which is one sixteenth of the original level, and this signal is fed to an adder 115 which is also supplied with a signal which is delayed one cycle by means of the delay circuit 117 and scaled by fifteen sixteenths by means of a second multiplier 119.
• the output signal from this part of the first-order recursive filter is hereby scaled by a certain factor, e.g. between one quarter and one sixteenth.
• the circuit is arranged as shown in the applicant's earlier European application no. 90309342.5.
• the • circuit is coupled to the algorithm control unit via the leads 82 and 83, so that the signal/noise ra- tio can be set by the algorithm control unit 79.
• the correlation circuit 31 and the FIR filter 27 are shown in detail in fig. 3 of the drawing.
• the FIR filter 27 is a standard digital filter of the type which comprises a delay line 41, a first multiplication amplifier 45 in front of the first delay stage 43, and a further multiplier 45 after each delay stage. All of the multipliers 45, each with its digital summing circuit 49, are connected in parallel.
• the digital signal on the delay line k thus passes a number of delay stages 43 in order to produce a series of sequen ⁇ tial signal samples x(n), x(n-l), x(n-2) ... etc., where x(n) is the latest digital example of the signal.
• Each sample is delayed one period controlled by the master clock which controls the A/D converter 17 and the D/A converter 19. It is typical for an all-in-the-ear aid for the upper frequency limit to be in the order of 7 kHz. This requires that the frequency of the master clock must be at least 14 kHz, and in practice at least 20 kHz.
• a master clock oscillator including a control ⁇ lable capacitor filter can be used, and can be preset to ⁇ produce a master clock frequency of either 10 kHz or 20 kHz.
• the FIR filter 27 is ar- ranged in a manner corresponding to the FIR filter in the applicant's earlier European application no. 90309342.5.
• the filter functions as follows:
• each of the coefficients h(m) is up- dated on each cycle of the master clock and a new output signal y(n) is calculated.
• Adaption is effected by a con ⁇ trolled adjustment of the value of the coefficient h(m).
• a correlation circuit 31 for this purpose is also shown in fig. 3.
• the correlator 31 is designed to adapt the filter 27 in accordance with the Widrow-Hoff algorithm (B. Widrow et al "Stationary and non-stationary learning character ⁇ istics of the LMS adaptive filter", Proc. IEEE volume 24 pages 1161 - 1162, August 1976).
• Each coefficient h(m) is adjusted every cycle, in that the adjustment is effected by increasing or decreasing the value of the coefficient, i.e. its magnitude and sign, which is carried out by the cor ⁇ relator 31.
• Each coefficient h(m) is stored independently in its own accumulator 59.
• the correlator 31 comprises a delay line 51 with a number of single-bit delay stages 53.
• the number of stages cor- responds to the number of stages 43 in the FIR filter 27.
• the input signal to the delay line 51 and the output signal from each delay stage 53 are coupled to the reference input of a digital multiplier stage 55.
• the second input to each multiplier stage 55 is coupled to a common set of digital' leads 39.
• the delay line 51 is coupled so that it receives the noise signal N from the noise source 33 and the delay" line 37, while the common set of digital leads 39 is con ⁇ nected to d in order to receive the residual signal.
• each multiplier stage 55 is coupled to an adap ⁇ tation-scale factor circuit 61, which via a summing circuit 57 feeds the signal to the coefficient accumulator 59.
• the circuit is arranged as ex ⁇ plained in the applicant's earlier European application no. 90309342.5.
• E(s) is the influence of the input amplitude
• E(noise) is the amplitude of the noise signal.
• Such an algorithm can be characterized as being an algo ⁇ rithm which provides a statistically reliable updating of the filter when the external feedback is constant.
• a hearing aid with such an algorithm will not be particu ⁇ larly quick to adapt itself to changes in the feedback path.
• the statistical probability of changes in the coefficients in the filter is known, i.e. when vari ⁇ ations take place in the number of filter coefficients which are undergoing change, it can hereby be ascertained when there is a significant change in the feedback path. For example, if it is determined that a significant change in the feedback path is involved when the coefficients in the filter exceed 4 x the standard deviation, a significant change has occurred in the acoustic feedback path.
• the circuit reacts by accelerating the adaptation, in that the insertion of more noise is ordered via the lead 82 and/or in another manner, e.g. by making ⁇ greater, an increased adaptation rate is ordered, whereby the adapta ⁇ tion circuit quickly brings the FIR filter to a state in which full compensation is achieved for the changes in the acoustic feedback path.
• the algorithm control circuit 79 determines that the coefficients are stable again, the noise level or the ⁇ -value is reduced and the feedback circuit again operates in accordance with the safe algorithm.
• a hearing aid with such a "double algorithm” will be cap ⁇ able of reacting considerably more quickly than the known aid according to the applicant's earlier European patent application no. 90309342.5, also even if 6 dB less noise is added in the statistically safe state, so that possible in- fluence on the user comfort can be further reduced.
• the hearing aid will function in a corresponding manner also if it is arranged to choose between more than two algorithms, merely providing that criteria are introduced into the circuit which determine under which conditions a decoupling takes place from the basic algorithm to one of the alternative algorithms.

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• Health & Medical Sciences (AREA)
• General Health & Medical Sciences (AREA)
• Neurosurgery (AREA)
• Otolaryngology (AREA)
• Physics & Mathematics (AREA)
• Engineering & Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
• Filters That Use Time-Delay Elements (AREA)
• Circuit For Audible Band Transducer (AREA)

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• 1992
  • 1992-03-31 DK DK43292A patent/DK170600B1/da not_active IP Right Cessation
• 1993
  • 1993-03-23 DE DE69327951T patent/DE69327951T2/de not_active Expired - Fee Related
  • 1993-03-23 WO PCT/DK1993/000106 patent/WO1993020668A1/en not_active Ceased
  • 1993-03-23 AU AU39482/93A patent/AU3948293A/en not_active Abandoned
  • 1993-03-23 JP JP5516993A patent/JPH07505271A/ja active Pending
  • 1993-03-23 EP EP93908828A patent/EP0634084B1/de not_active Expired - Lifetime

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