EP1118248A1 - Prothese auditive et procede de traitement de signaux de microphone dans une prothese auditive - Google Patents

Prothese auditive et procede de traitement de signaux de microphone dans une prothese auditive

Info

Publication number
EP1118248A1
EP1118248A1 EP99948785A EP99948785A EP1118248A1 EP 1118248 A1 EP1118248 A1 EP 1118248A1 EP 99948785 A EP99948785 A EP 99948785A EP 99948785 A EP99948785 A EP 99948785A EP 1118248 A1 EP1118248 A1 EP 1118248A1
Authority
EP
European Patent Office
Prior art keywords
signal
mic2
microphone signals
micl
hearing aid
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
EP99948785A
Other languages
German (de)
English (en)
Other versions
EP1118248B1 (fr
Inventor
Eghart Fischer
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sivantos GmbH
Original Assignee
Siemens Audioligische Technik GmbH
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Siemens Audioligische Technik GmbH filed Critical Siemens Audioligische Technik GmbH
Priority to DK99948785T priority Critical patent/DK1118248T3/da
Publication of EP1118248A1 publication Critical patent/EP1118248A1/fr
Application granted granted Critical
Publication of EP1118248B1 publication Critical patent/EP1118248B1/fr
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • 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 relates to a hearing aid with the features of the preamble of claim 1 and a method with the features of the preamble of claim 7.
  • the invention is intended for use with all types of hearing aids.
  • the invention is particularly suitable for highly developed hearing aids which, for example, have digital signal processing components.
  • a generic hearing aid is known from DE 43 27 901 Cl.
  • a signal processing unit serves to achieve a predetermined directional characteristic by a suitable mixture of signals from a plurality of microphones.
  • the properties of this directivity are fixed. Signal components from lateral signal sources are always damped and signal components from signal sources arranged in front of or behind the hearing aid wearer are amplified.
  • the object of the invention is to avoid the problems mentioned and to provide a hearing device and a method for processing microphone signals in a hearing device with high transmission quality and noise suppression in a large number of horse situations. According to the invention, this object is achieved by a hearing device with the features of claim 1 and a method with the features of claim 7.
  • the dependent claims relate to preferred embodiments of the invention.
  • the invention is based on the basic idea of varying the properties of an existing direction-dependent amplification / attenuation in accordance with the result of an additional signal analysis. This allows a particularly good adaptation of the hearing aid according to the invention to different horse situations. For example, the direction of a Storschall source can be taken into account in the direction-dependent amplification / damping, in order to offer good disturbance relief. If there is no significant noise, the attenuation can be switched off to minimize distortion.
  • the strengths of signal components of the microphone signals are determined in a plurality of predefined direction classes (W range) in the directional analysis. In this way, the rough direction of the main part of a Storschall source can be determined. Alternatively, provision can be made to determine the direction of one or more signal source (s) more precisely.
  • An adaptive LMS filter can be used for signal analysis, with which signal delays in particular are estimated by integer multiples of a sampling period.
  • the coefficients of the LMS filter determined by the adaptation process can influence or (completely) determine the result of the directional analysis or even represent this result.
  • different signal processing steps can be carried out in preferred embodiments.
  • the directional characteristic of a (directional microphone, formed by superimposing the microphone signals) can be suitably changed. Such a change can in particular be alignment of the directional microphone pole.
  • a suitable obstacle-free procedure can be selected.
  • Weighting signals are preferably generated during the evaluation of the signal analysis, which determine the weighting factors with which the results of different filtering, interference-free and / or straightening methods are received m the output signal.
  • the microphones for generating the microphone signals are arranged at a relatively small distance of at most 5 cm or at most 2.5 cm or approximately 1.6 cm from one another, the connecting line between the microphones being at an angle of at most 45 ° or at most Can extend 30 ° to the line of sight of the hearing aid wearer or can be approximately this line of sight.
  • a common housing can be provided for both microphones.
  • FIG. 2 shows a block diagram of a signal analysis unit in the circuit of FIG. 1, Fig. 3 em block diagram of an LMS filter m of the circuit of Fig. 2, and
  • Fig. 4 and Fig. 5 per em diagram of the change over time of coefficient signals or a microphone and an output signal m a signal example.
  • the hearing device circuit shown in FIG. 1 has a microphone unit 10 known per se, which contains two omnidirectional microphones 12, 12 'and a two-channel, equalizing preamplifier 14.
  • the two microphones 12, 12 ' are arranged at a distance of approximately 1.6 cm. This distance corresponds approximately to the distance that sound covers during a sampling period of the hearing aid circuit.
  • the connecting line between the two microphones 12, 12 ' runs approximately in the direction of view of the hearing aid wearer, the first microphone 12 being at the front and the second microphone 12' at the rear.
  • the microphone unit 10 generates first and second microphone signals MIC1, MIC2, which originate from the first and the second microphone 12, 12 ', respectively.
  • the two microphone signals MIC1 and MIC2 are fed to a signal analysis unit 16 and a signal processing unit 18.
  • the signal analysis unit 16 evaluates the microphone signals MIC1, MIC2 and generates three weighting signals G1, G2, G3 and e total weighting signal GG therefrom.
  • the signal processing unit 18 consists of a side signal reduction unit 20, a ratchet signal reduction unit 22 and a mixing unit 24.
  • the output signal OUT of the signal processing unit 18 is applied to a reproduction unit 26 and is there transmitted via an output amplifier 28 to a preferably electronic acoustic transducer 30, for example, a speaker.
  • the side signal reduction unit 20 receives the microphone signals MIC1, MIC2 and generates therefrom a first noise-reduced signal R1, in which signal components of the two microphone signals MIC1, MIC2, which originate from a sound source on the side of the hearing aid user, largely suppresses sm ⁇ .
  • the side signal reduction unit 20 has a subtractor 32, which forms the difference between the two microphone signals MIC1, MIC2.
  • the difference signal and the second microphone signal MIC2 are fed to a compensation unit 34 for generating the first noise-reduced signal R1.
  • the equalization unit 34 only forwards the difference signal received from the subtractor 32 as the first noise-reduced signal R1, the second microphone signal MIC2 not being taken into account.
  • the compensation unit 34 is designed as a predictor in order to achieve a better maintenance effect for signal components from lateral signal sources by suitable mixing of the difference signal and the second microphone signal MIC2.
  • a side signal reduction unit 20 with such a compensation unit 34 is described in the application of the same inventor with the title "Method for providing a directional microphone characteristic and hearing aid", the content of which is hereby incorporated into the present application.
  • the ratchet signal reduction unit 22 similar to the side signal reduction unit 20, has a subtractor 36 and a compensation unit 38, which generates a second noise-reduced signal R2. Those parts of the microphone signals MIC1, MIC2 that come from signal sources behind the hearing aid wearer are suppressed in the second noise-reduced signal R2.
  • the positive input of the subtractor 36 is connected to the first microphone signal MIC1, while the negative (to be subtracted) input is connected to the microphone signal MIC2 via the delay element 40, which causes a delay by one sampling period.
  • the signal reduction unit 22 can pass the compensation unit 38 unchanged on the difference signal of the subtractor 36 as a second noise-reduced signal R2.
  • the jerk signal reduction unit 22 can be provided with a compensation unit 38 designed as a predictor, as described in detail in the application mentioned in the preceding paragraph.
  • the mixing unit 24 has three weighting amplifiers 42, 44, 46, of which the first multiplies the first microphone signal MIC1 by the weighting signal G3, the second the first noise-reduced signal R1 by the weighting signal G2, and the third the second noise-reduced signal R2 by the weighting signal Eq.
  • the weighting signals G1, G2, G3 are thus used as gain values (gam values).
  • the output signals of the weighting amplifiers 42, 44, 46 are added by a summer 48.
  • the output signal of the summer 48 is multiplied by a further weighting amplifier 50 by the total weighting signal GG in order to obtain the output signal OUT of the mixing unit 24 (and the entire signal processing unit 18).
  • the more precise structure of the signal analysis unit 16 is shown in FIG. 2.
  • the filtered output signal Y of the LMS filter 52 is connected to the negative input of a subtractor 54.
  • the microphone signal MIC2 is present via the delay element 56, which provides a delay of three sampling periods, at the positive input of the subtractor 54, and the difference signal formed by the subtractor 54 is fed to the LMS filter 52 as an error signal E.
  • the following therefore applies for each sampling time t:
  • e (t) m ⁇ c2 (t-3) - y (t), (1) where e (t) is the error value of the error signal E at time t, y (t) the output value of the LMS filter 52 at time t and m ⁇ c2 (t-3) the value of the second microphone signal MIC2 at time t-3 (three clock cycles before time t).
  • a coefficient vector signal of the LMS filter 52 is present at a demultiplexer 58.
  • the coefficient vector signal transmits a coefficient vector w (t) which contains five values k ⁇ (t), kl (t), k2 (t), k3 (t), k4 (t) for the filter coefficients (taps) .
  • w (t) which contains five values k ⁇ (t), kl (t), k2 (t), k3 (t), k4 (t) for the filter coefficients (taps) .
  • the demultiplexer 58 determines five coefficient signals K0, Kl, K2, K3, K4 from the coefficient vector signal W, which indicate the value curve of the respective corresponding coefficient.
  • the three “middle” coefficient signals K 1, K 2, K 3 contain information about the spatial arrangement of the signal sources relative to the hearing aid wearer.
  • This assignment of the filter coefficients is the result of the delay of the second microphone signal MIC2 by three time units by the delay element 56.
  • the transmission of the coefficient vectors and the filter coefficients m to the coefficient vector signal W is carried out serially in the exemplary embodiment described here by means of a suitable protocol to which the Demultiplexer 58 is tuned. In the embodiment variants, the coefficients are transmitted in a different way, in particular in parallel or in part in parallel and in part in series.
  • a standardization unit 60 normalizes the three coefficient signals K1, K2, K3 and generates the weighting signals G1, G2, G3 and the total weighting signal GG therefrom.
  • Fig. 3 illustrates the internal structure of the LMS filter 52.
  • the input signal X is present at a buffer 62, which em Input vector signal Ü generated.
  • the input vector signal ⁇ expresses an input vector ü (t) which contains the values of the input signal X at the five preceding sampling instants. So the following applies:
  • x (t) indicates the value of the input signal X at the sampling time t.
  • the input vectors ü (t) are multiplied by a vector multiplier 64 in a matrix operation with the respective current coefficient vector w (t) of the coefficient vector signal W in order to obtain the (scalar) output value y (t) of the output signal Y at the clock instant t .
  • a vector multiplier 64 in a matrix operation with the respective current coefficient vector w (t) of the coefficient vector signal W in order to obtain the (scalar) output value y (t) of the output signal Y at the clock instant t .
  • FIR finite impulse response
  • An element square 66 generates the element-by-square of the signal vectors u (t), and an element summer 68 serves to sum up the squared elements. A small positive is added to the sum thus obtained by means of an adder 70
  • Constant C (order of magnitude 10 "10 ) added, which comes from a constant generator 72.
  • the result is as (scalar) Divisor on a scalar divider 74.
  • the dividend is the scalar product of the current error value e (t) of the error signal E and an output vector of a scalar multiplier 76. This output vector is produced by scalar multiplication of the input vector ü (t) with an adaptation constant ⁇ .
  • the result vector of the scalar divider 74 is added by a vector adder 78 to the current coefficient vector w (t).
  • E delay element 80 outputs the result only one clock pulse later than the adapted coefficient vector w (t + l) of the coefficient vector signal W.
  • w (t + i) w (t; ( ⁇ * e (t) * ü (t) / (C + ü (t) - ü ⁇ (t)))) (6)
  • the circuit shown in FIG. 3 implements an LMS algorithm which, by means of a stochastic gradient method, approximates (adapts) the filter coefficients k ⁇ (t) - k4 (t) in such a way that the error signal E is minimized as far as possible.
  • LMS algorithm which, by means of a stochastic gradient method, approximates (adapts) the filter coefficients k ⁇ (t) - k4 (t) in such a way that the error signal E is minimized as far as possible.
  • the first microphone 12 is located in front of the second microphone 12 ′ by about 1.6 cm in the direction of view of the hearing aid wearer. At a sampling frequency of 20 kHz assumed in the exemplary embodiment described here, this corresponds approximately to that
  • a signal SO from a useful sound source which is located in the direction of view of the hearing aid wearer (angle 0 °), will arrive at the front microphone 12 and at the sampling time t + 1 at the rear microphone 12 ', for example, due to the microphone spacing.
  • a signal S2 from a Storschall source which is located behind the hearing aid wearer (angle 180 °)
  • a signal S1 from a lateral noise source hits approximately simultaneously with both microphones 12, 12 'em and therefore also has an effect simultaneously on the microphone signals MIC1, MIC2.
  • m ⁇ cl (t) denotes the value of the signal MICl at the sampling time t.
  • e (t) is minimized by the algorithm of the LMS filter 52.
  • k3 (t) the term of which is the only one having the summand s0 (t-4) increases with increasing intensity of the signal SO (angle 0 °).
  • the amount of the filter coefficient k2 (t) is an indicator for the portion of the signal Sl (angle 90 °) m the microphone signals MIC1, MIC2, and the amount of the filter coefficient kl (t) shows the signal portion of S2 (angle) 180 °). The values of all other filter coefficients tend towards zero.
  • the weighting signals Gl, G2, G3 always correspond to the coefficient signals Kl, K2, K3.
  • differences in the weighting signals G1, G2, G3 can be enlarged ("spread").
  • the coefficient signals K1, K2, K3 serve directly as weighting factors.
  • the normalization unit 60 and the weighting amplifier 50 can then be omitted.
  • a large weighting factor Gl has the result that the second noise-reduced signal R2, in which the interference signal component is largely reduced from 180 °, receives a large proportion of the output signal OUT. Accordingly, with a large weighting factor G2, the first noise-reduced signal R1 largely influences the output signal OUT. With a large weighting factor G3, the first microphone signal MIC1 finally has a large effect on the output signal OUT.
  • the signal analysis unit determines the intensities or strengths of signal components of the microphone signals MIC1, MIC2 in the angular areas in the viewing direction of the hearing aid wearer, transversely to the viewing direction and behind the hearing aid wearer.
  • the weighting factors G1, G2, G3 correspond to the determined intensity values. Depending on these values, either signals from 90 ° or 180 ° are Signals are classified and largely suppressed, or the first microphone signal MIC1 is "switched through” if the directional analysis has determined that no significant (interference) signal components are present from either 90 ° or 180 °.
  • Kl (line - * - * -), K2 (line - + - + -) and K3 (line) in a realistic experiment with a useful signal source from 0 ° and an interference signal source from 90 ° (each voice signal).
  • the axis of abscissas represents the range from 0 to 10 seconds.
  • the value of the coefficient signal K2 (90 ° indicator) is always significantly higher than the value of the coefficient signal K1 (180 ° indicator).
  • the first microphone signal MIC1 and the output signal OUT for the signal example used in this experiment are shown in FIG. 5.
  • the microphone signal MIC1 mainly contains interference signal components. It can be seen that these components are largely suppressed in the output signal OUT.
  • the function of the hearing device and method according to the invention has been described with reference to the circuit shown by way of example in FIGS. 1 to 3, other implementations are possible in the alternative embodiments.
  • the functions of the circuit can be implemented in whole or in part by program modules of a digital processor, for example a digital signal processor.
  • the circuit can also be constructed as a digital or analog circuit or m different mixed forms between these extremes.
  • the result of the directional analysis is evaluated in another way for signal processing.
  • K2, K3 also for controlling time variants, for example three fixed directional microphone characteristics with poles at 90 °, 135 ° and 180 ° can be used.
  • design variants are provided in which an "intelligent" determination of interference and useful signal components is carried out (for example by means of the standardization unit 60). While in the embodiment described above, the signal • share (0 °) was always regarded as a useful signal component in the viewing direction, may for example in the presence of the signal Sl of 90 ° and simultaneous absence of the signal SO from 0 °, the signal Sl is now as Payload signal viewed and can no longer be suppressed.

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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)
  • Circuit For Audible Band Transducer (AREA)

Abstract

Prothèse auditive qui possède une unité de traitement (18) de signaux destinée à amplifier et/ou à atténuer en fonction de l'orientation des portions de signal provenant d'au moins deux signaux (MIC1, MIC2) de microphone. Ladite prothèse est en outre équipée d'une unité d'analyse (16) de signaux qui est en mesure de modifier une propriété de l'amplification ou de l'atténuation en fonction de l'orientation. Un procédé de traitement de signaux de microphone dans une prothèse auditive présente des caractéristiques correspondantes. La présente invention permet d'obtenir une prothèse auditive présentant une meilleure qualité de transmission et une meilleure atténuation des bruits parasites dans un grand nombre de situations d'écoute.
EP99948785A 1998-09-29 1999-09-17 Prothese auditive et procede de traitement de signaux de microphone dans une prothese auditive Expired - Lifetime EP1118248B1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
DK99948785T DK1118248T3 (da) 1998-09-29 1999-09-17 Höreapparat og fremgangsmåde til bearbejdning af mikrofonsignaler i et höreapparat

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
DE19844761 1998-09-29
DE19844761 1998-09-29
PCT/EP1999/006916 WO2000019770A1 (fr) 1998-09-29 1999-09-17 Prothese auditive et procede de traitement de signaux de microphone dans une prothese auditive

Publications (2)

Publication Number Publication Date
EP1118248A1 true EP1118248A1 (fr) 2001-07-25
EP1118248B1 EP1118248B1 (fr) 2005-03-23

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EP99948785A Expired - Lifetime EP1118248B1 (fr) 1998-09-29 1999-09-17 Prothese auditive et procede de traitement de signaux de microphone dans une prothese auditive

Country Status (4)

Country Link
US (1) US6751325B1 (fr)
EP (1) EP1118248B1 (fr)
DE (1) DE59911808D1 (fr)
WO (1) WO2000019770A1 (fr)

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Also Published As

Publication number Publication date
DE59911808D1 (de) 2005-04-28
WO2000019770A1 (fr) 2000-04-06
EP1118248B1 (fr) 2005-03-23
US6751325B1 (en) 2004-06-15

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