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EP0661905A2 - Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive - Google Patents

Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive Download PDF

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Publication number
EP0661905A2
EP0661905A2 EP95103571A EP95103571A EP0661905A2 EP 0661905 A2 EP0661905 A2 EP 0661905A2 EP 95103571 A EP95103571 A EP 95103571A EP 95103571 A EP95103571 A EP 95103571A EP 0661905 A2 EP0661905 A2 EP 0661905A2
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EP
European Patent Office
Prior art keywords
acoustic
individual
hearing aid
unit
loudness
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
EP95103571A
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German (de)
English (en)
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EP0661905A3 (fr
EP0661905B1 (fr
Inventor
Bohumir Dr. Sc.Techn. B.B.A. Uvacek
Herbert Dr. sc. tech. Bächler
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.)
Sonova Holding AG
Original Assignee
Phonak AG
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Filing date
Publication date
Application filed by Phonak AG filed Critical Phonak AG
Priority to EP01128611A priority Critical patent/EP1207718A3/fr
Priority to EP95103571A priority patent/EP0661905B1/fr
Priority to DK95103571T priority patent/DK0661905T3/da
Priority to AT95103571T priority patent/ATE229729T1/de
Priority to DE59510501T priority patent/DE59510501D1/de
Publication of EP0661905A2 publication Critical patent/EP0661905A2/fr
Publication of EP0661905A3 publication Critical patent/EP0661905A3/fr
Application granted granted Critical
Publication of EP0661905B1 publication Critical patent/EP0661905B1/fr
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    • 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
    • H04R25/70—Adaptation of deaf aid to hearing loss, e.g. initial electronic fitting
    • 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 present invention relates to a method according to the preamble of claim 1, a device according to that of claim 23 and a hearing aid according to claim 39.
  • a psycho-acoustic perception quantity is understood to be a quantity that is formed non-linearly, by individual laws of perception, from physical-acoustic quantities, such as frequency spectrum, sound pressure level, phase position, time course, etc.
  • Hearing aids known to date change physical, acoustic signal quantities in such a way that an hearing-impaired individual equipped with the hearing aid hears better.
  • the hearing aid is adapted by adjusting physical transmission variables, such as frequency-dependent amplification, level limitation, etc., until the individual is satisfied with the hearing aid within the scope of the possibilities presented.
  • Preferred embodiments of the method according to the invention are specified in claims 2 to 22, the device according to the invention in claims 24 to 38 and the hearing aid according to the invention in claim 40.
  • the device according to the invention can be designed as an adaptation device separately from the hearing device. However, it also includes adjustment measures on the hearing aid in order to correct the perceived size taken into account for the individual.
  • the loudness "L” is a psycho-acoustic quantity, which indicates how “loud” an individual feels a presented acoustic signal.
  • Loudness has its own unit of measurement; a sinusoidal signal with a frequency of 1 kHz and a sound pressure level of 40dB-SPL produces a loudness of 1 "Sone". A sine of the same frequency with a level of 50dB-SPL is perceived exactly twice as loud; the corresponding loudness is 2 sone.
  • the present invention has as its object to propose a method and devices suitable for this, with which a hearing aid to be adapted to an individual can be adjusted in such a way that the acoustic perception of the individual corresponds at least in a first approximation to that of a norm, namely the person with normal hearing.
  • FIG. 1 One possibility to record the individually perceived loudness on selected acoustic signals as a further usable variable is the one shown schematically in FIG. 1, for example from O. Heller, "Auditory field audiometry using the method of category division", Psychological Contributions 26, 1985, or V. Hohmann, "Dynamic Compression for Hearing Aids, Psychoacoustic Fundamentals and Algorithms", thesis UNI Göttingen, VDI-Verlag, series 17, No. 93, known method.
  • An individual I is presented with an acoustic signal A which can be adjusted on a generator 1 with regard to the spectral composition and transmitted sound pressure level S.
  • the individual I evaluates or "categorizes" the acoustic signal A currently heard according to e.g. thirteen loudness levels or categories, as shown in FIG. 1, to which levels numerical weights, for example from 0 to 12, are assigned.
  • the procedure according to the invention can also be used to take further psychoacoustic variables into account, such as for example the variable "masking behavior in the time domain and / or in the frequency domain".
  • the norm, N is used to determine a psycho-acoustic perception variable, in particular the loudness L N , by means of standardized acoustic signals A o and compared with the values of this variable, corresponding to L I of an individual, with the same acoustic signals A o . From the difference corresponding to ⁇ L NI , setting data are determined which act directly on a hearing aid or on the basis of which, manually, a hearing aid is set. L I is determined on the individual without a hearing aid or with a hearing aid that has not yet been adapted, possibly progressively adapted.
  • the loudness itself is a variable that in turn depends on several variables.
  • the correlation of detected size differences with interventions in the transmission behavior of a hearing aid is not clear and extremely complex.
  • a quantifying model of the Perception size, especially loudness used.
  • Such a model should be able to be entered with any kind of acoustic signals; at least approximate is the corresponding size.
  • the model that is valid for the individual should be identifiable with relatively few measurements. The identification should be able to be terminated when the model has been identified to a certain extent.
  • Such a quantifying model of a psycho-acoustic perception quantity does not have to be given by a closed mathematical expression, but can be defined by a multidimensional table, from which the perceived quantity of perception can be called up with the prevailing frequency and sound level relationships of a real acoustic signal as a variable can.
  • the band-specific, mean sound pressure levels S k form the model variables defining a presented acoustic signal, which determine the current spectral power density distribution.
  • the spectral width of the considered critical bands CB k , the linear approximation of the loudness perception, ⁇ k , and the hearing threshold T k are parameters of the model or the mathematical simulation function according to (1).
  • the model parameters ⁇ k , T k and CB k have been determined using the standard N, ie for people with normal hearing.
  • the curve L kN represents the loudness curve of the standard as a function of the sound level S k of an acoustic signal presented in a respective critical band k, recorded as explained with reference to FIG. 1.
  • a sinusoidal signal or a narrowband noise signal is presented.
  • the parameter ⁇ N represents the slope of a linear approximation or regression line of this course L kN at higher sound levels, ie at sound pressure levels from 40 to 120 dB SPL, where the acoustic useful signals also predominantly occur. This is also referred to below as "large signal behavior".
  • this increase can be assumed to be the same, ⁇ N , in each of the frequency bands.
  • the hearing threshold T kN In contrast to the parameter ⁇ N , the hearing threshold T kN also differs in the norm and in a first approximation in every critical frequency band CB kN and is not a priori identical to the 0dB sound pressure level.
  • the typical hearing threshold curve of the standard is precisely defined by ISO R226 (1961).
  • Leijon has described a procedure that allows the further band-specific coefficients or model parameters ⁇ kI and CB kI to be estimated from the hearing thresholds T kI of individuals.
  • the estimation errors are usually large when considering individual cases. Nevertheless, when identifying individual loudness models, it is possible to start with estimated parameters, for example those estimated from diagnostic information. This drastically reduces the effort and the burden on the individual.
  • the loudness L recorded with a category scaling according to FIG. 1, is plotted in FIG. 3 as a function of the mean sound pressure level in dB-SPL for a sinusoidal or narrow-band signal of the frequency f k in a critical band of the number k considered .
  • the loudness L N of the standard increases non-linearly with the signal level in the selected representation, the gradient curve is in a first approximation for normal hearing people for all critical bands with the regression line with the gradient ⁇ N entered on the curve N in FIG. 3 reproduced in [categories per dB-SPL].
  • model parameter ⁇ N corresponds to a nonlinear amplification, the same for normal hearing people in every critical band, but to be determined for individuals with ⁇ kI in every frequency band.
  • the straight line with the slope ⁇ k approximates the non-linear loudness function in band k by a regression line.
  • L kI typically denotes the course of the loudness L I of the hearing impaired in a band k.
  • the curve of a hearing impaired person has a larger offset to the zero point and is steeper than the curve of the norm.
  • the larger offset corresponds to an increased hearing threshold T kI
  • the phenomenon of the fundamentally steeper loudness curve is referred to as loudness recruitment and corresponds to an increased ⁇ parameter.
  • the width of the respective critical bands CB kI it can be stated that the presence of several such bands only becomes effective when psycho-acoustic processing of broadband audio signals, i.e. broadband signals, the spectrum of which is at least two adjacent critical bands. In hearing impaired people, a widening of the critical bands is typically noticeable, whereby primarily the loudness summation is impaired even after (1).
  • individual I as shown, for example, via headphones, electrically or by means of an electrical-acoustic transducer, is supplied with narrow-band norm-acoustic norm signals A ok lying in the frequency bands CB Nk .
  • the individual I evaluates and quantifies the perceived loudness, L S (A ok ).
  • the associated standard bandwidth CB kN and the parameter ⁇ N are provided on the output side via a selection unit 7 from a standard storage unit 9.
  • the electrical signal S e (A ok ) corresponding to the sound pressure level of the signal A ok is fed together with the associated bandwidth CB kN to a computing unit 11 which, according to the preferred one mathematical loudness model according to (1), a loudness value L '(A ok ) is calculated, namely from S e , CB kN , ⁇ N and, as mentioned previously, the hearing threshold value T kI stored in a storage unit 13 and predetermined .
  • loudness L 'the computing unit 11 calculates on the basis of these predetermined parameters. Based on the use of the hearing threshold T kI of the individual and the parameter ⁇ N of the standard, a loudness value L 'is determined on the computing unit 11 at the given sound level, corresponding to S e of the signal A ok , as it corresponds to a scaling function N', which is determined by the Regression line with ⁇ N and the hearing threshold T kI is defined in a first approximation.
  • this loudness value L ' is compared at a comparison unit 15 with the loudness value L I by the input unit 5.
  • the difference .DELTA. (L ', L I ) appearing on the output side of the comparison unit 15 acts on an incrementing unit 17.
  • the output of the incrementing unit 17 is superimposed on a superposition unit 19 with the ⁇ N parameter supplied to the computing unit 11 by the storage unit 9 with the correct sign.
  • the incrementing unit 17 thus increments the signal corresponding to ⁇ N by increments ⁇ according to the number of increments n until the difference appearing on the output side of the comparison unit 15 reaches or falls below a predeterminable minimum dimension.
  • the output signal of the comparison unit 15 in FIG. 4 is compared on a comparator unit 21 with an adjustable signal ⁇ r in accordance with a predeterminable, maximum error - as an abort criterion.
  • ⁇ r the difference signal
  • opening the switch Q1 and closing the switch Q2 on the one hand aborts the incrementation of ⁇
  • the parameter ⁇ kI of the individual is thus found with the required accuracy corresponding to ⁇ r in the critical frequency band k considered.
  • the process is optimally short or only as long as necessary.
  • Fig. 6a analogous to Fig. 5, the scaling function N of the norm and I of a hearing impaired individual is shown again.
  • an amplification G x must therefore be provided on the hearing device so that the individual perceives the loudness L x with the hearing device as the norm N.
  • 6a shows, depending on various, for example, entered sound pressure levels S kx , several gain values G x to be provided on the hearing aid are entered.
  • FIG. 6b shows the gain curve resulting from the considerations of FIG. 6a as a function of S k , as can be realized on a transmission channel on the hearing aid corresponding to the critical frequency band k, as shown in FIG. 6c.
  • the non-linear gain curve G k (S k ) shown in FIG. 6b is determined heuristically and schematically from the parameters T kI and ⁇ kI or the differences T kN -T kI and n ⁇ as determined with reference to FIGS. 4 and 5.
  • the described procedure is optimally repeated in every critical frequency band k. For each critical frequency band and when approximating with a regression line, only one norm-acoustic signal has to be presented to the individual; if necessary, others can be used to check the regression lines found.
  • the model according to (1) which is preferably used becomes arbitrarily more precise (1 *) by using ⁇ k (S k ) instead of the level-independent parameters ⁇ k .
  • ⁇ k is replaced by ⁇ k (S k ).
  • FIG. 8 shows the scaling curve N of the norm and of an individual I in analogy to FIG. 5.
  • the scaling curve N is sound pressure level-dependent slope parameter ⁇ N (S k) is approximated, ie by a polygon of support values S k of the curve N.
  • This sound pressure level dependent parameter ⁇ N (S k) are assumed to be known by they can be easily determined from the known scaling curves N of the standard at the given support values S kx .
  • a set of sound pressure level-dependent slope parameters ⁇ N (S k ) is stored in the memory unit 9.
  • the individual I is again presented with normacoustic, narrow-band signals lying in the respective critical bands, but, in contrast to the procedure according to FIG. 4, per critical frequency band at different sound pressure levels S kx .
  • the individual loudness evaluations for these standard acoustic signals of different sound pressure levels are preferably stored in a buffer unit 6. With reference to FIG. 8, these stored loudness perception values hold the scaling curve I of the individual with reference values.
  • the storage unit 9 supplies the bandwidth CB kN associated with the critical frequency band under consideration and the set of ⁇ parameters dependent on sound pressure level to the computing unit 11, in addition to the previously determined, individual, band-specific hearing threshold T kI .
  • the frequency of the norm acoustic signal determines the critical frequency band k under consideration, and the values relevant for this are retrieved from the memory unit 9 accordingly.
  • the sequence F of the following sound pressure level values S kx is preferably further stored in a memory device 10. As soon as the individual loudness perception values are recorded and stored in the storage unit 6, the sequence of the stored sound pressure level values S kx is also fed from the storage unit 10 to the computing unit 11, with which the latter, according to FIG.
  • the difference appearing on the output side of the comparison unit 15, here in the sense of a difference in sound pressure level dependent between the curves S and the changed curve N 'according to FIG. 8, is assessed with regard to falling below a predetermined maximum range - as a termination criterion - and as soon as the deviations mentioned indicate a TARGET 4, on the one hand the optimization or incrementation process is interrupted; on the other hand, the sound pressure level-dependent ⁇ parameters pending at the computing unit 11 are output which correspond to the tangent slope values on the individual scaling curve I, i.e. ⁇ kI (S kx ) or the ⁇ ' ⁇ kI (S kx ).
  • the nonlinear amplification function assigned to the specific critical frequency band on the hearing aid is determined and adjusted there.
  • the width of the critical bands CB k becomes relevant for the loudness perception of the individual if the presented normacoustic signals have spectra that lie in two or more critical frequency bands, because loudness summation according to (1) or (1 *) then occurs .
  • frequency bands CB k and CB k + 1 for example critical frequencies for the standard N, are drawn in over the frequency axis f.
  • frequency bands CB k and CB k + 1 for example critical frequencies for the standard N, are drawn in over the frequency axis f.
  • the nonlinear reinforcements found so far have been determined channel-specifically or band-specifically with reference to the critical bandwidths of the standard. Taking into account the critical bandwidths of the individual, it can be seen from FIG. 9a that, for example, the hatched area ⁇ f in the individual falls within the broadened critical band k, while in the norm it falls within the band k + 1. However, this means that, with the previous reference to the critical bandwidths of the standard, signals e.g. in the hatched frequency range .DELTA.f, the gain must be corrected on the individual.
  • FIG. 10 shows a further development as a function block signal flow diagram in which the parameters ⁇ k and CB k can be determined using a single method. Not only is one critical band after the other examined in accordance with FIGS. 4 and 7, but also, with broadband acoustic signals, the loudness summation is recorded and the width of the individual critical bands is thus also determined as a variable by optimization.
  • the simulation model parameters of the standard namely ⁇ N , CB kN , are stored in a memory unit 41 and, in a preferred embodiment, not the hearing thresholds T kN of the standard, but rather the hearing thresholds T kI of the individual to be examined, determined beforehand by audiometry and taken from a memory unit 43.
  • An individual is acoustically presented with signals A ⁇ k by a generator that is no longer shown here.
  • the electrical signals corresponding to them in FIG. 10, also designated A ⁇ k are fed to a frequency-selective power measurement unit 45.
  • the channel-specific average powers are determined on the unit 45 in accordance with the critical frequency bands of the standard, frequency-selective, and a set of such power values S ⁇ k is output on the output side.
  • These signals are stored in a memory unit 47 in a channel-specific manner and specifically for the signal A ⁇ k (A No.) that is presented in each case.
  • an arithmetic module 53 calculates the loudness L 'according to (1) from the norm parameters ⁇ N , CB kN and the individual hearing threshold values T kI , taking into account the loudness summation , which would result for the norm if the latter had hearing thresholds (T kI ) such as the individual.
  • the calculated value L ' N is stored in a storage unit 55 on the output side of the computing module 53.
  • Each of the presented broadband ( ⁇ k) signals A ⁇ k is assessed or categorized by the individual in terms of loudness perception, the evaluation signal L I , again assigned to the respective presented acoustic signals A ⁇ k , stored in a storage unit 57. Both when determining L ' N and when determining L I , the loudness summation is taken into account arithmetically or by the individual due to the broadbandness ⁇ k of the signals A ⁇ k presented.
  • the corresponding number of values L ' N is stored in the memory unit 55, and the corresponding number of L I values are also stored in the memory unit 57.
  • the parameter modification unit 49 varies the start values ⁇ N , CB kN , but not the T kI values, for all critical frequency bands, while simultaneously recalculating the updated L ' N value until the difference signal ⁇ (L' N , L I ) runs within a predeterminable minimum course, which is checked on the unit 61.
  • the standard parameters ⁇ N and CB kN entered as start values taking into account the signals S ⁇ k corresponding to the channel-specific sound pressure values retrieved from memory 47, are varied according to predetermined search algorithms until a maximum permissible deviation between the L ' N and the L I course has been reached.
  • ⁇ and CB values on the output side of the modification unit 49 correspond to those which, used in (1), result in loudness values corresponding optimally with the individually perceived values L I for the acoustic signals A ⁇ k presented: by varying the standard parameters, the individual values in turn became individual determined.
  • manipulated variables are determined in order to set the amplification functions on the frequency-selective channels of the hearing aid corresponding to the critical frequency bands.
  • the procedure described is actually the search for a minimum point of a multi-variable function.
  • several sets of changed parameters will result in the minimum criterion specified by ⁇ R being met.
  • the method described can therefore lead to the receipt of several such solution parameter sets, with the physical positions of the hearing aid being used in those sets which are physically sensible and, for example, the easiest to implement.
  • Solution parameter sets that can be excluded from the outset which would lead, for example, to amplification profiles on the respective channels of the hearing aid that are extremely difficult or impossible to implement, can be excluded from the outset by corresponding specifications on the modification unit 49.
  • a shortening of the search process can also be achieved, for example for hearing-impaired individuals, by replacing the standard parameters ⁇ N and CB kN with the ⁇ kI and CB kI values estimated from the individual hearing thresholds T kI for hearing impaired people as search starting values in the Storage unit 41 are stored, especially if the individual's hearing loss is determined from the outset.
  • the arithmetic unit 51 can also do the mentioned Include storage devices integrated in terms of hardware; their delimitation shown in dashed lines in FIG. 10 is to be understood, for example, including in particular the computing module 53 and the coefficient modification unit 49.
  • the previously described procedure according to FIGS. 4, 7 and 10 are primarily suitable for setting a hearing aid ex situ.
  • the determined manipulated variables can be transferred directly electronically to a hearing device in situ, but the actual advantage of an in situ adaptation, namely the consideration of the fundamental hearing influence by a hearing device, is not taken into account: first, all the manipulated variables are determined without a hearing device, and then is set without further acoustic signal presentation.
  • the acoustic signals A ⁇ k are fed to the hearing aid system HG with transducers 63 and 65 on the input and output sides and individual I, the latter loading the perceived L I values into the memory 57 with the evaluation unit 5.
  • the L I value is stored in the memory 57 for each presented standard-acoustic, broadband signal A ⁇ k .
  • the loudness values L ' N are initially determined on the computing module 53 according to (1) or (1 *), as was explained with reference to FIG. 10 , calculated and, specifically assigned to the presented signals A ⁇ k , stored in the memory unit 55.
  • the standard parameters from the memory unit 41 are then modified, as described, until they, when used in (1) or (1 *), give L ' N values with predeterminable accuracy corresponding to the L I values in memory 57.
  • L ' N L I. for all A. ⁇ k .
  • the hearing aid HG has a number k o frequency-selective transmission channels K between the converter 63 and converter 65.
  • Actuators for the transmission behavior of the channels are connected to an actuating unit 70 via a corresponding interface. The latter are fed the initial manipulated variables SG o previously determined as optimal.
  • the parameter changes found will act ⁇ ⁇ k , ⁇ ⁇ CB k , ⁇ ⁇ T k or Parameters ⁇ N , T kN , CB kN and ⁇ kI , T kI , CB kI via the manipulated variable control unit 70 so as to control the hearing aid in such a way that its channel-specific frequency and amplitude transmission behavior for the signals A ⁇ k , on the output side, produce the correction loudness L Kor .
  • the control variable determination unit 70 according to FIG. 11, manipulated variable changes, is summarized from the parameter changes determined in FIG. 11 in analogy to the procedure according to FIG. 10, in order to convert N 'into I according to FIG. 8, and from the differences in the hearing thresholds ⁇ SG for the channel-specific frequency and amplitude transmission behavior of the hearing aid in such a way that the scaling curves of the individual I are brought with the hearing aid HG with the desired accuracy to the scaling curves N of the standard:
  • the loudness behavior of the hearing aid forms the intrinsic, i.e.
  • the individual's "own” loudness perception depends on that of the norm, the loudness perception of the individual with a hearing aid becomes equal to that of the norm or can be specified in relation to that of the norm.
  • the "in situ” setting shown, for example, with reference to FIG. 11 shows the essential The advantage is that the physical “in situ” transmission behavior of the hearing aid and, for example, the mechanical ear influencing by the hearing aid are also taken into account.
  • the hearing aid should, when optimally set, transmit received acoustic signals with the correction loudness L Kor to its output, so that the system hearing aid and individual has a perception that is equal to that of the standard or ( ⁇ L in Fig. 12a) deviates from this by a predeterminable amount.
  • channels 1 to k o are provided on a hearing aid according to the invention, followed by an acoustic-electrical input converter 63, each assigned to a critical frequency band CB kN .
  • the entirety of these transmission channels forms the signal transmission unit of the hearing aid.
  • the frequency selectivity for channels 1 to k o is implemented by filter 64.
  • Each channel also has a signal processing unit 66, for example with multipliers or programmable amplifiers.
  • the non-linear, band- or channel-specific amplifications described above are implemented on the units 66.
  • the converted acoustic input signals present on the output side of the converter 63 are converted into their frequency spectrum at a unit 64a.
  • the aforementioned channel-specific correction parameters and the corresponding correction loudness L KOR are converted into actuating signals SG66 on the computing unit 53 ', with which the units 66 are set.
  • the values .DELTA.SG supplied to the hearing aid according to FIG. 12a) according to FIG. 11 therefore essentially correspond to the channel-specific correction parameters in this embodiment variant.
  • the hearing aid transmits the input signals mentioned with the correction loudness L KOR .
  • the system individual with hearing aid thus perceives the required loudness, be it preferably the same as the standard or in this respect in a predetermined ratio.
  • the spectra are formed from the converted acoustic input signals and the electrical output signals of the hearing aid at units 64a.
  • a computing unit 53a are due of the input spectra as well as the loudness model parameters of the norm N calculates the current loudness values which the norm would perceive on the basis of the input signals.
  • the loudness values that the individual perceives without the hearing aid, ie the intrinsic individual are calculated on a computing unit 53b on the basis of the output signal spectra.
  • the modeling parameters 53b are fed with the modeling parameters of the individual, which, as described above, were determined.
  • a controller 116 compares the loudness values L N and L I determined by standard and individual modeling and, channel-specifically, the parameters of the standard model and the individual model and, on the output side, sends control signals SG66 to the transmission units 66 in accordance with the determined differences in such a way that the modeled Loudness L I becomes equal to the currently required standard loudness L N.
  • controller 116 In contrast to the correction model variant of FIG. 12a), controller 116 first determines the necessary correction loudness L KOR in accordance with FIG. 12b).
  • the hearing aid transmission is set with the units 66 in such a way that the acoustic signals currently pending are transmitted with the correction loudness, so that the loudness is modeled on the output signals in accordance with the perception behavior of the individual ( 53b), results in a loudness corresponding to that which is perceived by the standard or which can be specified in this respect.
  • FIG. 13 An embodiment of a hearing aid according to the invention, combined from the procedure according to FIG. 11 and the structure according to FIG. 12a), is shown in FIG. 13.
  • the same position signs as in FIGS. 11 and 12 are used for the same function blocks.
  • only one channel X of the hearing aid is shown.
  • a switchover unit 81 connects the storage unit (41, 43, 44) according to FIG. 11, shown here as a unit, to the unit 49.
  • a switchover unit 80 is in the position shown, i.e. is open, a switchover unit 84 is also initially effective in the position shown.
  • the arrangement works exactly as shown in FIG. 11 and explained in this context.
  • the determined parameter changes ⁇ k , ⁇ CB k , ⁇ T k , which convert the individual loudness model (I) into the standard loudness model (N), when the hearing aid is put into operation by switching over the switching unit 80 in the storage unit 41 ', 43', 44 'acting in the same way as the storage unit 41, 43, 44 is loaded.
  • the switching unit 81 is switched to the output of the last-mentioned storage unit.
  • the modification unit 49 is deactivated (DIS), so that it directly supplies the data from the storage unit 41 'to 44' unmodified and permanently to the computing unit 53c.
  • the switchover unit 84 is switched over so that the output on the arithmetic unit 53c, now acting as arithmetic unit 53 'according to FIG. 12a), via the manipulated variable control unit 70a onto the transmission path with the units 66 of the Hearing aid works.
  • the ⁇ Z k parameters ⁇ k , ⁇ CB k , ⁇ T k act together with L KOR on the manipulated variable control unit 70a.
  • the loudness model arithmetic unit 53c integrated in the hearing aid is initially used to determine the model parameter changes ⁇ k , ⁇ CB k , ⁇ T k required for correction and then, in operation, to guide the transmission manipulated variables of the hearing aid in a time-variable manner - in accordance with the current acoustic signals Relationships - used.
  • correction loudness model parameters on the hearing aid and thus the necessary manipulated variables for generally non-linear channel-specific amplifications, e.g. for a hearing impaired person, allows different target functions, or the loudness requirements as a target function, as mentioned, can be achieved with different sets of correction loudness model parameters and therefore manipulated variables ⁇ SG66.
  • Fig. 14 shows the measures to be taken in addition to the precautions of Fig. 11; the same function blocks, which have already been listed in FIG. 11 and thus explained, have the same item numbers.
  • a sound sensation structured according to specific categories can also be numerically scaled, for example according to the criteria known from Nielsen. 14 and 11, after hearing device HG has been set by finding a correction parameter set ( ⁇ k , ⁇ CB k , ⁇ T k ) such that the individual with the hearing device has at least approximately the same loudness perception as the norm, the individual states: for example, in the case of the same broadband norm-acoustic signals A ⁇ k presented , on a sound scaling unit 90. A numerical value is assigned to each sound category on the unit 90.
  • the individually quantified sound sensation KL I with the, for example, statistically determined sound sensation KL N becomes the norm for the same acoustic Signals A ⁇ k compared. These are stored in a memory unit 94 so that they can be called up.
  • a sound characterization unit 96 for example between the comparison unit 59 and the parameter modification or incrementation unit 49, is activated according to FIG. 14, which limits the degree of freedom of the parameter modification on the unit 49 , ie changes one or more of the parameters mentioned, regardless of the minimal difference obtained at unit 59, and keeps them constant.
  • the sound characterization unit 96 is preferably connected to an expert database, shown schematically in FIG. 14 at 98, which contains the information regarding individual Sound sensation deviation from the norm is supplied. Information, for example, is stored in the expert database 98 "shrill at A ⁇ k is the result of too much amplification in channels no. "
  • the gain in one or more of the higher-frequency hearing aid channels is withdrawn, with which the termination criterion ⁇ R according to FIG. 10 is no longer met at the comparison unit 59 and a new search cycle for uses the correction model parameters, but with the retraction of the gain in higher-frequency hearing aid channels prescribed by the expert system.
  • a specific constellation of simultaneously prevailing correction coefficients ⁇ k , ⁇ CB k and ⁇ T k in a critical frequency band k can be regarded as a band-specific state vector Z k ( ⁇ k , ⁇ CB k , ⁇ T k ) of the correction loudness model.
  • the entirety of all band-specific state vectors Z k forms the band-specific state space, which is three-dimensional in the case considered here.
  • Band-specific state vectors Z k are primarily responsible for every sound feature that can occur during sound scaling, with "shrill and" muffled "in high-frequency critical bands. This expert knowledge must be stored as rules in the sound characterization unit 96 or the expert system 98.
  • a changed state vector Z ' k must be searched for in at least one of the critical bands in order to change the sound.
  • the parameters of the correction loudness model on the hearing device thus result, based on the parameters of the standard, from a first incremental change “ ⁇ ” for conforming loudness adjustment and from second incremental changes ⁇ for sound matching.
  • FIG. 15 again in a functional block representation, the hearing aid according to the invention according to FIG. 12b) (model difference variant) is shown in a form which is preferably implemented.
  • the same reference numerals are used as were used for the hearing aid according to the invention according to FIG. 12b).
  • the output signal of the input converter 63 of the hearing aid is subjected to a time / frequency transformation at a transformation unit TFT 110.
  • the resulting signal in the frequency domain, is transmitted in the multi-channel time-variant loudness filter unit 112 with the channels 66 to the frequency / time domain FTT transformation unit 114 and from there, in the time domain, to the output converter 65, for example a loudspeaker or another stimulus transducer for the Individual.
  • the loudness L N is calculated according to the input signal in the frequency domain and the standard model parameters in accordance with Z kN .
  • the individual loudness L I is calculated analogously on the output side of the loudness filter 112.
  • the loudness values L N and L I are supplied to the controller unit 116.
  • the individual loudness is corrected to the standard loudness by adapting the isophones of an individual to those of the standard.
  • the intelligibility of speech is not yet optimal. This stems from the masking behavior of the human ear, which is different in the case of damaged individual hearing than in the norm.
  • the frequency masking phenomenon states that quiet tones in close frequency neighborhood are faded out from loud tones, i.e. they do not contribute to loudness perception.
  • FIG. 16 shows, based on the representation of the hearing aid according to the invention according to FIG. 15 described so far, a further development in which, in addition to the loudness correction of the individual, a masking correction for a hearing-impaired individual, and hence frequency demasking, is carried out. It should be noted in advance that changing the masking behavior of the hearing aid and thus its frequency transmission behavior also changes the loudness transmission, which means that the loudness transmission must be iteratively recreated after each change in the frequency masking behavior.
  • the input signal of the hearing aid is supplied in the frequency domain to a standard masking model unit 118a, whereupon the input signal is masked as in the standard. How the masking model is determined will be explained later.
  • the output signal of the hearing aid in the frequency range is, analogously, supplied to the individual masking model unit 118b, whereupon the output signal of the hearing aid is subjected to the masking model of the intrinsic individual.
  • the input and output signals masked with the models N and 1 are fed to the masking controller 122 and compared there. In the function of the comparison results, the controller 122 intervenes in a regulating manner on a masking filter 124 until the masking "hearing aid transmission and individual "is aligned with that of the norm.
  • the multichannel time-variable loudness filter 112 is followed by the likewise multichannel time-variable masking filter 124, which, as mentioned, is set in function of the difference determined at the masking controller 122 such that the norm-masked input signal at unit 118a equals the "individual + hearing aid" -masked output signal at unit 118b will. If the transmission behavior of the hearing aid has now been changed via the masking controller 122 and the masking filter unit 124, the correction loudness L KOR of the transmission no longer corresponds to the required one, and the loudness controller 116 adjusts the manipulated variables on the multi-channel time-variable loudness filter 112, that the controller 116 again determines the same loudness L I , L N.
  • Masking correction via controller 122 and loudness tracking via controller 116 are thus carried out iteratively, the loudness model used, defined by the state vectors Z LN , Z LI , remaining unchanged. It is only when both the loudness controller 116 and the masking controller 122 that the iterative matching of the filters 112 and 124 achieves the same within narrow tolerances, is the transmitted signal at the frequency / time transformation unit 114 converted back into the time domain and transmitted to the individual .
  • the frequency masking model is parameterized by state vectors Z FMN or Z FMI .
  • a static acoustic signal is presented to the human ear, for example with the three frequency components f 1-f 3 shown, a masking curve F fx is assigned to each frequency component according to its loudness.
  • the norm perceives a loudness to which the unmasked components L f1N -L f3N contribute.
  • the slopes m unN and m obN of the masking curves F f are essentially independent of frequency and level if, as shown, the frequency scaling takes place in "bark", according to E. Zwicker (in critical bands).
  • the masking curves F f are broadened as far as the gradients m are concerned, and they are also raised. This can be seen from the illustration for a hearing-impaired individual I below in Fig. 17, according to which, with the same acoustic signals presented with the frequency components f1-f3, the component on the frequency f2 is not perceived and thus also contributes to the perceived loudness.
  • the frequency masking behavior of the standard N is again shown in dashed lines in characteristic I of FIG. 17.
  • the total masking limit FMG formed by all frequency-specific masking characteristic curves F f naturally also varies over the entire frequency spectrum, with which the filter 126 or the channel-specific filter must be guided in a time-variable manner.
  • the frequency masking model for the standard is known from E. Zwicker or from ISO / MPEG according to the literature reference below.
  • the applicable individual frequency masking model with FMG I must first be determined in order to be able to carry out the individually necessary correction, as shown schematically with the unmasking filter 126 in FIG. 17.
  • frequency components which are masked according to the frequency masking model of the standard i.e. which do not contribute to loudness, are not taken into account at all, i.e. not broadcast.
  • Narrow band noise R o preferably centered with respect to the center frequency f o of a critical frequency band CB k of the standard or, if already determined as described above, the individual, is presented to the individual via headphones or, and preferably, via the already loudness-optimized hearing aid.
  • a sinusoidal signal preferably at the center frequency f o , is added to the noise R o , as are sinusoidal signals at f un and f ob above and below the noise spectrum. These test sinus signals are added sequentially in time. By varying the amplitude of the signals to f un , f o and f ob , it is determined when the individual to whom the noise R o is presented perceives a change in this noise.
  • the corresponding perception limits determine three points of the frequency masking behavior F foI of the individual.
  • certain estimates are preferably used in advance in order to shorten the investigation process.
  • the masking at the center frequency f o is initially estimated to be -6dB for the hearing impaired.
  • the frequencies f un and f ob are chosen to be offset by one to three critical bandwidths with respect to f o . This procedure is preferably carried out at two to three different center frequencies f o , distributed over the hearing range of the individual, in order to determine FMG I , the frequency masking model of the individual or its parameters, such as in particular m obf , m unf .
  • FIG. 19 schematically shows the experimental setup for determining the frequency masking behavior of an individual according to FIG. 18.
  • Noise center frequency f o , noise bandwidth B and the average noise power A N are set on a noise generator 128.
  • the output signal of the noise generator 128 is superimposed with the respective test sinus signals, which are set on a sine generator 132.
  • Amplitude A S , frequency f S can be set on the test sine generator 132.
  • the test sine generator 132 is preferably operated in a clocked manner, for which purpose it is activated cyclically, for example via a clock generator 134.
  • the superimposition signal is fed to the individual via an amplifier 136 via calibrated headphones or, and preferably, directly via the hearing aid according to FIG. 16, which is still to be optimized with respect to frequency masking.
  • the noise signals R o are presented to the individual, for example every second, and the respective test sinusoidal signal TS is added to one of the noise packets.
  • the individual is asked whether and, if so, which of the noise packages sounds different from the others. If all noise packets sound the same to the individual, the amplitude of the test signal TS is increased until the corresponding noise packet is perceived differently from the others, then the associated point A W is found on the frequency masking characteristic FMG I according to FIG. 18.
  • the unmasking model according to block 126 of FIG. 17 can be determined from the masking model of the individual determined in this way and the known standard.
  • the TARGET masking is actually calculated at block 118a depending on the acoustic signal presented, and the filter 124 in the signal transmission path is adjusted via the masking controller 122 until the masking on it and on the individual - model on 118b - provides the same result as required by the leadership masking model in block 118a.
  • the frequency masking correction generally also changes the loudness transmission, so that loudness control and frequency masking control are carried out alternately until both criteria are met with the required accuracy, only then is the "virtually present" acoustic signal converted back into the time range via block 114 and transmitted to the individual.
  • the frequency masking behavior says that if there is a spectral component of an acoustic signal with high level, simultaneous spectral components with low levels and in close frequency neighborhood of the high level component may not contribute to the perceived loudness, it follows from the masking behavior in time that after a loud acoustic signal there may be no noise. That is why slower speaking is helpful for unmasking the time of a hearing impaired person.
  • the frequency / time inverse transformation unit 114 (Wigner inverse transformation or Wigner synthesis) is preceded by a spectrum / time buffer 142 which acts analogously to the buffer 140.
  • a further computing device 53 ′ b determines the time image of the L I values determined on the basis of the spectra. This time image is compared with the time image of the L N values at controller 116a, and the comparison result is used to control a multi-channel loudness filter unit 112a with controlled, time-variable dispersion (phase shift, time delay).
  • the filter 112a thus ensures that the temporal correction loudness image of the transmission with the loudness image of the individual corresponds to that of the norm.
  • the 142 respectively stored spectra in the buffers 140, the total of signals over a predetermined time period, for example from 20 to 100 msec depict, time and frequency masking model computers for the standard 118 'a and the individual 118' are further b supplied to the are parameterized with the norm and individual parameters or state vectors, Z FM , Z TM . Both frequency masking model F N , analogous to FIG. 16, and time masking model T M are implemented therein.
  • the outputs of the computers 118 ' a , 118' b act on a masking controller unit 122a, the latter acting on the multi-channel unmasking filter 124a, which can now also be used to control the dispersion in a time-variable manner in addition to 124 from FIG. 16.
  • the control of the loudness filter 112a and the masking correction filter 124a is preferably carried out alternately, until both assigned controllers 116a and 122a detect predetermined minimum deviation criteria. Only then are the spectra in the buffer unit 142 converted back into the time domain in the correct time sequence at the unit 114 and transmitted to the individual wearing the hearing aid.
  • Fig. 21 shows a hearing aid structure in which loudness correction, frequency masking correction and time masking correction on signals converted into the frequency range.
  • a technically possibly simpler embodiment variant according to FIG. 22 consistently takes into account time phenomena on signals in the time domain and phenomena with regard to frequency response on signals in the frequency domain.
  • a time masking correction unit 141 is connected upstream of the time / frequency transformation unit 110, which according to the embodiment of FIG. 16 preferably carries out an instantaneous spectrum transformation, as shown schematically, or, if necessary also as a supplement or replacement, between reverse transformation unit 114 and output transducer 65, such as speakers, stimulator, e.g. an electrode-stimulated cochlear implant.
  • the signal processing in block 117 takes place in accordance with the processing between 110 and 114 of FIG. 16.
  • the time mask correction unit designated 140 in FIG. 22 is shown in more detail in FIG. 23. It comprises a time-loudness model unit 142, by means of which, preferably as a power integral, the course of the loudness is tracked over the time of the acoustic input signal. Analogously, the instantaneous loudness of the signal in the time domain before its conversion is determined at the time / frequency transformation unit 110 in a further time-loudness model unit 142.
  • the loudness curves in the time of the mentioned input signal and the mentioned output signal are compared on a (simplified) time-loudness controller 144, and on a filter unit 146, namely essentially a gain control unit GK, the loudness of the output signal over which Time considered, that of the input signal aligned.
  • the input signal is fed to a time buffer unit 148, which, according to W. Verheist, M. Roelands, “An overlap-add technique based on waveform similarity ", ICASSP 93, pp. 554-557, 1993, WSOLA Algorithms or, according to E. Moulines, F. Charpentier, "Pitch Synchronous Waveform Processing Techniques for Text to Speech Synthesis Using Diphones", Speech Communication Vol. 9 (5/6), pp. 453-467, 1990, PSOLA- Algorithms are used.
  • a standard time masking model unit 150 N the standard time masking to be described is modeled on the input signals, on the further unit 150 I , on the output signals of the time buffer unit 148, the individual time masking.
  • the time maskings modeled on the signals on the input and output sides of the time buffer unit 148 are compared on a time masking control unit 152, and according to the comparison result, the signal output on the time buffer unit 148 is time-controlled via the algorithms mentioned, preferably used, ie the transmission via the time buffer 148 controlled time-variable expansion factor or delay.
  • the time masking behavior of the standard is again known from E. Zwicker.
  • the time masking behavior of an individual will be explained with reference to FIG. 24.
  • time masking limit curve ZMG for example of a hearing-impaired individual
  • I the time masking limit curve
  • TMG N the dot-time masking behavior
  • TMG N the dot-time masking behavior of the curve N
  • a time masking correction basically involves either delaying the second signal A2 on the individual - with the hearing aid - until his individual time masking limit has fallen sufficiently far, or the signal A2 so too reinforce that the individual is also above his time masking limit.
  • the last-mentioned procedure on the individual shows that A2 has to be strengthened so that in the best case the same perceived area L is above the time masking limit of the individual.
  • the decay time T AN of the time masking limit TMG N on the Norm is essentially independent of the level or loudness of the signal triggering the time masking, as shown in Fig. 24 of A1. This also applies to people with hearing loss, so that in most cases it is sufficient to determine the decay time T AI of the time masking limit TMG I regardless of the level.
  • the individual time masking limit decay time T AI 25 to determine the individual time masking limit decay time T AI, the individual is presented with a click-free and click-free exposing narrowband noise signal R o . After exposure of the noise signal R o a test sinusoidal signal with Gaussian wrap-around him will be presented after a set interval T Paus. A point corresponding to A ZM of the individual time masking limit TMG I is determined by varying the envelope amplitude and / or the pause time T Paus . Further changes in the pause time and / or the envelope amplitude of the test signal determine two or more points of the individual time masking limit.
  • test sine generator 132 which emits a Gauss-encased sine signal. The individual is asked at which pair of values T Paus and amplitude of the Gauss envelope the test signal after the noise signal is currently being perceived.
  • the individual masking behavior can also be estimated from diagnostic data, which results in a significant reduction in the time for the identification of the individual time masking model TMG I.
  • the essential parameter of this model is the decay time T AN or T AI .

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  • Acoustics & Sound (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Measurement Of The Respiration, Hearing Ability, Form, And Blood Characteristics Of Living Organisms (AREA)
  • Measurement Of Mechanical Vibrations Or Ultrasonic Waves (AREA)
  • Adornments (AREA)
  • Measurement Of Velocity Or Position Using Acoustic Or Ultrasonic Waves (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Auxiliary Devices For And Details Of Packaging Control (AREA)
EP95103571A 1995-03-13 1995-03-13 Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive Expired - Lifetime EP0661905B1 (fr)

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EP01128611A EP1207718A3 (fr) 1995-03-13 1995-03-13 Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive
EP95103571A EP0661905B1 (fr) 1995-03-13 1995-03-13 Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive
DK95103571T DK0661905T3 (da) 1995-03-13 1995-03-13 Fremgangsmåde til tilpasnning af et høreapparat, anordning hertil og høreapparat
AT95103571T ATE229729T1 (de) 1995-03-13 1995-03-13 Verfahren zur anpassung eines hörgerätes, vorrichtung hierzu und hörgerät
DE59510501T DE59510501D1 (de) 1995-03-13 1995-03-13 Verfahren zur Anpassung eines Hörgerätes, Vorrichtung hierzu und Hörgerät

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EP1207718A2 (fr) 2002-05-22
DK0661905T3 (da) 2003-04-07
EP0661905A3 (fr) 1995-10-04
EP1207718A3 (fr) 2003-02-05
EP0661905B1 (fr) 2002-12-11
DE59510501D1 (de) 2003-01-23

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