EP0661905B1 - 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
EP0661905B1
EP0661905B1 EP95103571A EP95103571A EP0661905B1 EP 0661905 B1 EP0661905 B1 EP 0661905B1 EP 95103571 A EP95103571 A EP 95103571A EP 95103571 A EP95103571 A EP 95103571A EP 0661905 B1 EP0661905 B1 EP 0661905B1
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EP
European Patent Office
Prior art keywords
acoustic
individual
unit
hearing
loudness
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EP95103571A
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German (de)
English (en)
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EP0661905A3 (fr
EP0661905A2 (fr
Inventor
Bohumir Dr. Sc.Techn. B.B.A. Uvacek
Herbert Dr. sc. tech. Bächler
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Sonova Holding AG
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Phonak AG
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Priority to AT95103571T priority Critical patent/ATE229729T1/de
Priority to DE59510501T priority patent/DE59510501D1/de
Priority to DK95103571T priority patent/DK0661905T3/da
Priority to EP01128611A priority patent/EP1207718A3/fr
Priority to EP95103571A priority patent/EP0661905B1/fr
Publication of EP0661905A2 publication Critical patent/EP0661905A2/fr
Publication of EP0661905A3 publication Critical patent/EP0661905A3/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 The preamble of claim 1, an apparatus according to that of claim 23 and a hearing aid according to claim 39.
  • a psycho-acoustic perceptual quantity becomes one Size understood, which is non-linear, through individual laws perception, from physical-acoustic Variables such as frequency spectrum, sound pressure level, phase position, Course of time, etc.
  • Hearing aids known to date change physical, acoustic Signal sizes such that a hearing aid hearing impaired individual hears better.
  • the adaptation the hearing aid is done by setting physical Transmission quantities, such as frequency-dependent amplification, Level limitation etc. until the individual with the Satisfied hearing aid within the possibilities presented is.
  • Preferred embodiment variants of the method according to the invention are specified in claims 2 to 22 of the invention Device in claims 24 to 38 and of the hearing aid according to the invention in claim 40.
  • the device according to the invention can designed as a fitting device separately from the hearing aid his. However, it also includes adjustment measures on the hearing aid the perceived size taken into account for the individual correct.
  • inventive device defined in the claims, the inventive method and the inventive Hearing aids are then, for example, based on Figures explained.
  • the loudness "L” is a psycho-acoustic quantity, which indicates how “loud” an individual is at presented acoustic Signal senses.
  • Loudness has its own unit of measurement; a sinusoidal 1kHz frequency signal at a sound pressure level of 40dB-SPL, produces a loudness of 1 "Sone". A sine of same frequency with a level of 50dB-SPL is exactly double perceived so loud; the corresponding loudness is so 2 sone.
  • the present invention has as its object propose a method and suitable devices for this, with which a hearing aid to be adapted to an individual can be adjusted so that the acoustic perception of the individual at least in the first approximation of that one Norm, namely the normal hearing.
  • the procedure according to the invention can certainly also for the consideration of other psychoacoustic Sizes are used, such as for the consideration of the size "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, which in turn is of depends on several variables.
  • this is the number Measurements that must be made on an individual to get even approximate information, with the interventions on the hearing aid, for everyone in a natural environment occurring broadband signals that are desired To be able to correct perception, great.
  • the correlation between recorded size differences Interventions in the transmission behavior of a hearing aid ambiguous and extremely complex.
  • a quantifying model of the Perception size, especially loudness used.
  • a model is intended to be used with all types of acoustic signals can be received; at least approximate results the corresponding size sought.
  • the model can be identified, that is valid for the individual. The identification should be able to be canceled if the model is in predeterminable Extent is identified.
  • Such a quantifying model of a psycho-acoustic Perception size does not have to be closed mathematical expression can be given, but may well be defined by a multidimensional table from where with the prevailing frequency and sound level ratios a real acoustic signal as a variable the perceived size can be called up.
  • 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 mathematical loudness model according to (1), calculates a loudness value L '(A ok ) , namely from S e , CB kN , ⁇ N and, as previously mentioned, predetermined hearing threshold value T kI stored in a memory unit 13.
  • 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.
  • 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, depending on various, for example, entered sound pressure levels S kx , a plurality of amplification 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 and the differences T kN -T kI and n ⁇ as determined with reference to FIGS. 4 and 5.
  • the described procedure is optimally used in every critical one Frequency band k repeated. It has to be critical Frequency band and approximation with a regression line only presented a norm-acoustic signal to the individual become; more can be checked if necessary of the regression lines found are used.
  • 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 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 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.
  • the partially broadened, corresponding bands are entered for an individual I.
  • the nonlinear reinforcements found so far have been channel-specific or band-specific with reference to the critical ones Bandwidths of the standard determined.
  • the critical bandwidths of the individual is from Fig. 9a it can be seen that, for example, the hatched area ⁇ f in the individual falls within the broadened critical band k, while in the norm it falls in the band k + 1. This means but that, with the previous reference to the critical bandwidths the standard, signals e.g. in the hatched frequency range ⁇ f in the individual must be corrected for gain.
  • 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.
  • the computing 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 storage unit 55, as is the corresponding number L I values in the storage 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.
  • Control variables are determined from the parameter values present on the output side of the modification unit 49 when the search is terminated and their difference from the start values ⁇ N and CB kN in order to set the amplification functions on the frequency-selective channels of the hearing aid corresponding to the critical frequency bands.
  • Solution parameter sets excluded from the outset can be, for example, only extremely difficult or unrealizable gain curves on the respective Channels of the hearing aid could lead through appropriate Specifications on the modification unit 49 from the outset be excluded.
  • a shortening of the search process can also be achieved, for example for hearing-impaired individuals, by replacing the standard parameters ⁇ N or CB kN with the ⁇ kI or 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 hearing loss of the individual is determined from the outset.
  • the arithmetic unit 51 can also do the mentioned Include storage devices integrated in terms of hardware; is its delimitation shown in dashed lines in FIG. 10 to understand, 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 the setting of a hearing aid ex situ.
  • the determined manipulated variables may well be direct electronically transmitted to a hearing aid in situ, whereby but the real benefit of in situ adjustment, namely the consideration of the fundamental hearing impairment through a hearing aid, is not considered: First all manipulated variables are determined without a hearing aid, and then, without further acoustic signal presentation, its Setting made.
  • 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. .delta..sub.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 changed parameters ⁇ ' Nk , CB' Nk have been determined for a predetermined number of presented normacoustic, broadband signals A ⁇ k by means of the computing module 53 and the modification unit 49, by means of which, according to FIG. 8, the Scaling curves N 'have been adapted to those of the individual I with a hearing aid HG that has not yet been adjusted, the parameter changes found act ⁇ ⁇ k , ⁇ ⁇ CB k , ⁇ ⁇ T k or the parameters ⁇ N , T kN , CB kN and ⁇ kI , T kI , CB kI via the manipulated variable control unit 70 in such a way that it controls 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 loudness behavior of the hearing aid forms the intrinsic, i.e. "own” loudness perception of the individual that of the norm, the loudness perception of the individual with hearing aid becomes or is the same as that of the norm, based on that of the standard, can be specified.
  • Fig. 12a) and b) are two basic implementation variants of a hearing aid according to the invention, using simplified signal flow function block diagrams, which are "ex situ”, but preferably “in situ” as described can be put.
  • 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.
  • All signal processing units 66 act on the output side to a summation unit 68, which in turn is on the output side to the electrical-acoustic output transducer 65 of the hearing aid acts. Until then, the two versions are correct according to FIGS. 12a) and 12b).
  • 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 SG 66 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.
  • 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 SG 66 to the transmission units 66 in accordance with the determined differences, such 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).
  • FIG. 11 An embodiment of a hearing aid according to the invention, combined from the procedure according to FIG. 11 and the structure 12a) is shown in FIG. It is for the same Function blocks have the same position symbols as in Fig. 11 and 12 used. For reasons of clarity, only one Channel X of the hearing aid shown.
  • Switching unit 81 according to the storage unit (41, 43, 44) Fig. 11, shown here as a unit, with the unit 49.
  • a switching 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), acts via the manipulated variable control unit 70a on the transmission path with the units 66 of the hearing aid.
  • 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.
  • the determination of the correction loudness model parameters on the hearing aid and thus the necessary manipulated variables for generally non-linear channel-specific amplifications, e.g. for the hearing impaired, allows different target functions, or the loudness requirements can be used as a target function, as mentioned, with different sets of correction loudness model parameters and therefore manipulated variables ⁇ SG 66 can be achieved.
  • the hearing aid optimally set the gain frequency selective, i.e. in certain transmission channels, raised, the correction loudness changes.
  • FIG. 14 shows that in addition to the precautions of FIG. 11 measures to be taken; the same functional blocks which already 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 is compared with the sound sensation KL N of the norm, for example, which is statistically determined for the same acoustic signals A ⁇ k . These are stored in a memory unit 94 so that they can be called up.
  • 14 becomes a sound characterization unit according to FIG 96, for example between comparison unit 59 and parameter modification incrementing unit 49, activated, which the parameter modification on the unit 49 in limited in their degree of freedom, i.e. one or more of the mentioned parameters, regardless of the minimum at unit 59 received difference, changed and constant.
  • the sound characterization unit 96 is preferably connected to an expert database, shown schematically at 98 in FIG. 14, to which the information relating to individual sound sensitivity 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. ."
  • 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.
  • band-specific correction state vectors Z k which give the individual a sense of loudness with the hearing aid essentially the same as that of the standard, as described above, have been found, then a changed state vector Z ' k must be sought in at least one of the critical bands to change the sound.
  • a changed state vector Z ' k When changing the one band-specific state vector, it must either be changed further so that the loudness remains the same, or at least one other band-specific state vector must also be changed.
  • 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. 12b again in functional block representation, is Hearing aid according to the invention according to FIG. 12b) (model difference variant) presented in a form as is preferred is realized. To make the overview easier the same reference numerals used as for the hearing aid according to FIG. 12b) were used.
  • 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 standard loudness L N is calculated from 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 corrected to the standard loudness by the isophones of an individual are brought into line with those of the norm.
  • the objective function "standard loudness" and possibly also achieved sound perception optimization language is understandable not yet optimal. This is due to the masking behavior of human hearing, which in a damaged individual hearing is different from the norm.
  • the frequency masking phenomenon states that soft tones in close frequency neighborhood of loud tones faded out will not contribute to loudness perception.
  • the intelligibility is to be further increased, then it must ensure that those spectral components, that are unmasked in the standard, i.e. perceived, also if individual hearing is damaged are perceived, the latter mostly through a distinguishes widened masking behavior. With the injured Hearing components were usually masked, which are unmasked in the standard hearing.
  • the input signal of the hearing aid is in the frequency range supplied to a standard masking model unit 118a, where the input signal is masked as with the Standard. How the masking model is determined will be shown later explained.
  • the output signal of the hearing aid in the frequency domain is analog, supplied to the individual masking model unit 118b, whereupon the output signal of the hearing aid the masking model of the intrinsic individual.
  • the input and output signals masked with the N and I models are supplied to the masking controller 122 and compared it. In function of the comparison results Controller 122 accesses a masking filter in a regulatory sense 124 until the mask "hearing aid transmission and individual "are aligned with those of the norm is.
  • 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 to the individual transfer.
  • the frequency masking model is parameterized by state vectors Z FMN or Z FMI .
  • a masking curve F fx is assigned to each frequency component in accordance with its loudness. Only the level components that exceed the masking limits, corresponding to the F f functions, contribute to the sound and loudness perception of the broadband signal presented, for example with the frequency components f 1 -f 3 .
  • 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.
  • 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 according to the frequency masking model of the norm be masked, so don't contribute to loudness at all not taken into account, 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 on a superposition unit 130 with the respective test sinusoidal 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 regard 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 at block 118a calculated according to the acoustic signal presented and, via masking controller 122, filter 124 in FIG Signal transmission path adjusted until the masking on it and on the individual - model on 118b - the same Result delivers, as from the leadership masking model in block 118a required.
  • changes with frequency masking correction generally also the loudness transmission, so that loudness control and frequency masking control alternately until both are made Only then will criteria be met with the required accuracy via block 114, the "quasi currently" acoustic signal is present Signal converted back into the time domain and the individual transmitted.
  • the frequency / time inverse transformation unit 114 (Wigner inverse transformation or Wigner synthesis) is an analog to Buffer 140 acting spectrum / time buffer 142 upstream.
  • 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.
  • Driving 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. First then the spectra in the buffer unit 142 are correct Time sequence on unit 114 converted back into the time domain and transmitted to the individual wearing the hearing aid.
  • 21 shows a hearing device structure in the case of loudness correction, Frequency masking correction and time masking correction on signals converted into the frequency range.
  • a technically possibly simpler design variant 22 consistently takes time phenomena in signals into account in the time domain and phenomena related to frequency response Signals in the frequency domain. This is done before the time / frequency transformation unit 110, which according to the execution 16 preferably shows an instantaneous spectrum transformation executes a time mask correction unit, as shown schematically 141 upstream 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 time mask correction unit designated 140 in FIG. 22 is shown in more detail in FIG. it includes a time-loudness model unit 142 on which, preferably as a performance integral, the course of the loudness over the Time of the acoustic input signal is tracked. Analogous is in another time-loudness model unit 142 instantaneous loudness of the signal in the time range before it Conversion determined at the time / frequency transformation unit 110.
  • the loudness curves in the time of the input signal mentioned and the output signal mentioned are on compared to 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, considered over time, that of the input signal equalized.
  • the input signal is used to carry out the time masking correction fed to a time buffer unit 148, according to which W. Verhelst, 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 used become.
  • 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 in accordance with 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 from E. Zwicker known.
  • the time masking behavior of an individual is to be explained with reference to FIG. 24.
  • a second acoustic signal A 2 which is subsequently presented, is only perceived if its level is above the time masking limit TMG N shown in broken lines.
  • TMG N time masking limit
  • FIG. 24 shows the time masking limit profile ZMG of, for example, a hearing-impaired individual under representation I with the same, schematically represented acoustic signals A 1 and A 2 . It can be seen that the second in the time signal A 2 is not perceptible when the hearing impaired may.
  • the dot-time masking behavior TMG N assumed for example, of the curve N is again shown in dash-dotted lines in the course of I. From the difference it can be seen that a time masking correction basically involves either delaying the second signal A 2 on the individual - using the hearing aid - until his individual time masking limit has dropped sufficiently, or the signal A 2 to be strengthened in such a way that the individual is also above his time masking limit.
  • the perceived area of the signal A 2 is designated L in the course of N, the last-mentioned procedure on the individual reveals that A 2 must be amplified so that, in the best case, the same perceived area L is above the individual's time masking limit.
  • the decay time T at the time masking TMG limit N to the standard is essentially independent of the level or loudness of the time masking triggering signal, as shown in FIG. 24 of A 1. This also applies to hearing impaired people, 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 narrow-band 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 .

Claims (40)

  1. Procédé d'adaptation d'une prothèse auditive (HG) à un individu (I), dans lequel
    on quantifie au moins une grandeur de perception psycho-acoustique (L, Ff) d'une norme (N) à des signaux acoustiques donnés ;
    on quantifie la même grandeur de perception psycho-acoustique (L, Ff) telle que l'individu (I) la perçoit aux signaux acoustiques donnés ;
    on règle ou on conçoit à partir d'écarts des grandeurs de perception psycho-acoustiques quantifiées mentionnées la prothèse auditive pour l'individu de façon que la grandeur de perception psycho-acoustique, telle que perçue par l'individu avec la prothèse auditive, soit au moins proche de celle, perçue par la norme, en une relation prédéfinissable,
    caractérisé en ce qu'on établit pour la dépendance de la grandeur de perception psycho-acoustique (L, Ff) de signaux acoustiques d'allures spectrales quelconques un modèle à plusieurs paramètres, paramétré avec au moins une partie des bandes de fréquence critiques (CBk) et en ce qu'on règle ou conçoit, au moyen des différences déterminées par les quantifications mentionnées des paramètres au modèle de la norme et au modèle de l'individu, la prothèse auditive pour l'individu.
  2. Procédé selon la revendication 1, caractérisé en ce que la relation prédéfinissable est l'égalité.
  3. Procédé selon l'une des revendications 1 ou 2,
    caractérisé en ce qu'on procède aux quantifications, la détermination des écarts avec un dispositif séparé de la prothèse auditive et qu'on présente les signaux acoustiques à l'individu sans prothèse auditive pour la quantification.
  4. Procédé selon l'une des revendications 1 ou 2,
    caractérisé en ce qu'on effectue la quantification, la détermination des écarts avec un dispositif séparé de la prothèse auditive et qu'on présente les signaux acoustiques à l'individu avec la prothèse auditive pour la quantification et qu'on établit de préférence entre le dispositif et la prothèse auditive une connexion pouvant être commandée pour la transmission de données qui dépendent des écarts.
  5. Procédé selon l'une des revendications 1 à 4,
    caractérisé en ce qu'on arrête la quantification des grandeurs de perception psycho-acoustiques par l'individu lorsque les écarts sont déterminés avec une précision prédéfinissable (ΔR).
  6. Procédé selon l'une des revendications 1 à 5,
    caractérisé en ce qu'on réduit le nombre des grandeurs à quantifier par l'individu en ce qu'on estime au préalable sa perception, de préférence sur la base d'informations diagnostiques et qu'on vérifie l'estimation par la quantification et qu'on la précise le cas échéant.
  7. Procédé selon l'une des revendications 1 à 6,
    caractérisé en ce qu'on utilise comme grandeur de perception psycho-acoustique au moins la sonie ou le masquage des fréquences.
  8. Procédé selon l'une des revendications 1 à 7,
    caractérisé en ce que, pour la détermination de la dépendance de la grandeur de perception psycho-acoustique de signaux acoustiques, on définit les paramètres de modèle de façon que la grandeur psycho-acoustique modelée sur la base des signaux acoustiques est perçue d'une manière égale à celle perçue par la norme sur la base des signaux acoustiques mentionnés, en ce qu'on quantifie ensuite la grandeur psycho-acoustique perçue par l'individu sans prothèse auditive à des signaux acoustiques (5) et qu'on modifie les paramètres de modèle au modèle de façon que la grandeur psycho-acoustique modelée formée d'une manière calculée coïncide selon une mesure prédéfinissable avec celle quantifiée par l'individu.
  9. Procédé selon l'une des revendications 1 à 8,
    caractérisé en ce qu'on arrête la détermination des paramètres pour le modelage de la grandeur perçue par l'individu lorsque les paramètres déterminent le modèle avec une précision prédéfinissable.
  10. Procédé selon l'une des revendications 1 à 9,
    caractérisé en ce que la définition des paramètres commence par des valeurs d'estimations de ceux-ci.
  11. Procédé selon l'une des revendications 1 à 10,
    caractérisé en ce qu'on détermine uniquement des paramètres qui déterminent la modélisation avec une précision prédéfinissable.
  12. Procédé selon l'une des revendications 1 à 11,
    caractérisé en ce qu'on met en oeuvre à la prothèse auditive le modèle (53' ; 118, 120 ; 53a, 118a ; 150) et qu'on fixe les paramètres de celui-ci pour former un modèle de correction conformément aux différences respectivement modifications mentionnées.
  13. Procédé selon l'une des revendications 1 à 11,
    caractérisé en ce qu'on met en ouvre à la prothèse auditive le modèle pour la norme et pour l'individu, qu'on applique respectivement à des signaux d'entrée et de sortie de la prothèse auditive et qu'on règle la transmission de la prothèse auditive en fonction des différences de modelage.
  14. Procédé selon l'une des revendications 1 à 13,
    caractérisé en ce qu'on sélectionne un modèle (1) où les modifications des paramètres (α, CB, T) produisent les mêmes modifications de la grandeur psycho-acoustique modélisée que des modifications de grandeurs de positionnement physiques associées (66) produisent des modifications de la grandeur psycho-acoustique au trajet de transmission à la prothèse auditive.
  15. Procédé selon l'une des revendications 1 à 14,
    caractérisé en ce que plusieurs jeux de modifications de paramètres, qui répondent aux conditions indiquées, sont définis, et que le jeu pour la conception ou le réglage de la prothèse auditive ou le guidage de sa transmission est utilisé qui produit pour l'individu avec la prothèse auditive une impression sonore individuellement satisfaisante.
  16. Procédé selon l'une des revendications 1 à 15,
    caractérisé en ce qu'on utilise comme grandeur de perception psycho-acoustique la sonie et que celle-ci est modelé par
    Figure 00910001
    où :
    k :
    paramètre de marche avec 1 ≤ k ≤ ko, numérotation du nombre ko de bandes critiques considérées ;
    CBk :
    largeur spectrale de la bande critique considérée avec le numéro k ;
    αk :
    montée d'une approximation linéaire de la sensation de la sonie gradué en catégories lors de l'application logarithmique du niveau d'un signal acoustique présenté sinusoïdal ou à bande étroite dont la fréquence se situe à peu près au milieu de la bande critique considérée CBk ;
    Tk :
    seuil d'audition au signal sinusoïdal mentionné ;
    Sk :
    le niveau de pression sonore moyen d'un signal acoustique présenté dans la bande de fréquence critique considérée CBk ;
    et où le cas échéant le modèle est élargi pour des αk dépendant du niveau.
  17. Procédé selon la revendication 16, caractérisé en ce que lors du modelage de l'individu, les seuils d'audition sont considérés individuellement, de préférence également les αk et le cas échéant également la CBk sont considérés individuellement.
  18. Procédé selon l'une des revendications 1 à 16,
    caractérisé en ce que le modèle est paramétré individuellement avec au moins une partie des bandes de fréquence critiques.
  19. Procédé selon l'une des revendications 16 ou 17,
    caractérisé en ce que le masquage de fréquence et/ou de temps est utilisé en plus comme grandeur de perception psycho-acoustique.
  20. Procédé selon l'une des revendications 1 à 19 ,
    caractérisé en ce qu'on modélise la dépendance de signaux acoustiques d'une grandeur psycho-acoustique à la prothèse auditive pour la norme et pour un individu et qu'on applique les modèles à des signaux électriques d'entrée et/ou de sortie, correspondant aux signaux acoustiques de la prothèse auditive dans la plage de temps et/ou dans la plage de fréquence.
  21. Procédé selon la revendication 19, caractérisé en ce qu'on utilise par intermittence à la prothèse auditive au moins un modèle de sonie et au moins un modèle de masquage pour le guidage de grandeurs de positionnement de transmission.
  22. Procédé selon l'une des revendications 1 à 21,
    caractérisé en ce qu'on utilise comme grandeur de perception psycho-acoustique le masquage du temps et qu'on tient compte de celui-ci à la prothèse auditive avec un retard de transmission variable dans le temps d'une manière commandée, de préférence en utilisant des algorithmes WSOLA.
  23. Dispositif pour l'adaptation d'une prothèse auditive à un individu avec au moins une unité de calcul (11 ;53, 53' ; 118, 120 ; 53a, 118a ; 150), où au moins un modèle (L, Ff, ZMG) est mis en ouvre, qui modélise la dépendance d'une grandeur de perception psycho-acoustique de l'homme de signaux acoustiques et avec laquelle, côté entrée, est reliée fonctionnellement une entrée pour des signaux dépendant de signaux acoustiques, caractérisé en ce qu'une unité de comparaison (15 ; 59 ; 116 ; 122 ; 116a, 122a ; 152) est prévue dont l'entrée est fonctionnellement reliée à la sortie de l'unité de calcul et qui présente une entrée supplémentaire qui peut être reliée fonctionnellement à une entrée pour l'entrée d'une grandeur de perception psycho-acoustique quantifiée, où la sortie de l'unité de comparaison émet des signaux pour la conception ou pour le réglage ou pour le guidage du comportement en transmission de la prothèse auditive.
  24. Dispositif selon la revendication 23, caractérisé en ce qu'il est connecté, côté entrée, à l'unité de calcul une unité de stockage avec des données fixes et que la sortie de l'unité de comparaison agit sur une entrée de commande d'une unité de modification de données, à laquelle les données acheminées par l'unité de stockage à l'unité de calcul sont modifiées en fonction du signal à la sortie de l'unité de comparaison.
  25. Dispositif selon la revendication 24, caractérisé en ce que la sortie de l'unité de comparaison agit sur une unité de valeurs de seuil dont la sortie active respectivement arrête l'unité de modification, où est amené à l'unité de valeurs de seuil un signal de valeur de seuil prédéfinissable.
  26. Dispositif selon l'une des revendications 23 à 25,
    caractérisé en ce que le dispositif à la prothèse auditive comprend au moins une unité de calcul qui est reliée, côté entrée, à une unité de stockage et à laquelle sont acheminés des signaux en fonction des signaux d'entrée et/ou de sortie de la prothèse auditive , où l'unité de calcul agit, côté sortie, sur des organes de positionnement pour la transmission à la prothèse auditive.
  27. Dispositif selon la revendication 26, caractérisé en ce que sont amenés à l'unité de calcul à la fois des signaux d'entrée et de sortie et en ce qu'agissent sur les organes de positionnement des signaux en fonction d'une différence du signal de sortie de l'unité de calcul, obtenu respectivement avec les signaux d'entrée respectivement de sortie.
  28. Dispositif selon l'une des revendications 22 à 27,
    caractérisé en ce qu'est mis en ouvre à l'unité de calcul au moins un modèle qui modélise au moins l'une des grandeurs de perception psycho-acoustique, sonie, masquage de fréquences, masquage de temps, qui modélise de préférence au moins la sonie.
  29. Dispositif selon la revendication 28, caractérisé en ce qu'une unité de calcul est prévue d'une manière décalée de la prothèse auditive sur laquelle agit, côté entrée, par une installation de modification de données une unité de stockage pour des données fixes, où l'unité de comparaison agit, côté sortie, sur une entrée de commande à l'unité de modification de données et où est prévu en outre un générateur de signaux qui agit, d'une part, sur une entrée de commande d'émission à l'unité de stockage, d'autre part sur un convertisseur électrique/acoustique, où l'unité de calcul modélise une grandeur psycho-acoustique, paramétrée avec des données modifiées acheminées par l'unité de stockage.
  30. Dispositif selon la revendication 29, caractérisé en ce que l'unité de comparaison est fonctionnellement reliée, côté entrée, à une unité de graduation de catégorie à laquelle peut être catégorisée individuellement la perception.
  31. Dispositif selon la revendication 28, caractérisé en ce qu'il est prévu à la prothèse auditive au moins une unité de calcul à laquelle le modèle est mis en ouvre, et en ce qu'il est associé à celle-ci une unité de stockage pour des données de paramètre, où elle agit, côté sortie, sur des organes de positionnement pour la transmission des signaux à la prothèse auditive.
  32. Dispositif selon la revendication 31, caractérisé en ce que sont stockés à l'unité de stockage au moins deux jeux de données qui agissent sur l'unité de calcul respectivement avec les signaux d'entrée et de sortie de la prothèse auditive, que la différence de modélisation est formée à ceux-ci en fonction de laquelle l'unité de calcul agit sur les organes de positionnement.
  33. Dispositif selon l'une des revendications 23 à 32,
    caractérisé en ce qu'il est mis en ouvre à au moins une unité de calcul précitée un modèle de sonie selon
    Figure 00960001
    où :
    k :
    paramètre de fonctionnement avec 1 ≤ k ≤ ko, numérotation du nombre ko de bandes critiques considérées ;
    CBk :
    largeur spectrale de la bande critique considérée portant le numéro k ;
    αk :
    montée d'une approximation linéaire de la sensation de la sonie graduée en catégories logarithmique du niveau d'un signal acoustique présenté sinusoïdal ou à bande étroite dont la fréquence se situe à peu près au milieu de la bande critique considérée CBk ;
    Tk :
    seuil d'audition au signal sinusoïdal mentionné ;
    Sk :
    le niveau de pression sonore moyen d'un signal acoustique présenté dans la bande de fréquence critique considérée CBk;
    et où le cas échéant le modèle mis en ouvre tient compte de la dépendance du niveau de αk.
  34. Dispositif selon l'une des revendication 23 à 33,
    caractérisé en ce qu'il est monté en amont des deux entrées de l'unité de comparaison une unité de stockage intermédiaire (55, 57).
  35. Dispositif selon l'une des revendications 24 à 34,
    caractérisé en ce qu'il est amené à l'unité de calcul une entrée pour des signaux acoustiques par une unité de formation de puissance (45, 47).
  36. Dispositif selon la revendication 26, caractérisé en ce que le trajet de transmission (117) à la prothèse auditive est disposé entre une unité de transformation de plage de temps en plage de fréquences (110) et une unité de transformation (114) de plage de fréquences en plage de temps, et en ce que l'unité de calcul est fonctionnellement reliée à l'entrée et à la sortie du trajet de transmission.
  37. Dispositif selon la revendication 36, caractérisé en ce qu'un trajet de transmission supplémentaire (148) est prévu en amont de l'unité de transformation de plage de temps en plage de fréquences (110) et qu'une unité de calcul (150) est fonctionnellement reliée, côté entrée, à la fois à l'entrée et aussi à la sortie du trajet de transmission supplémentaire (148) et effectue des modélisations à l'aide des signaux de sortie et d'entrée du trajet de transmission supplémentaire (148) où une unité de comparaison (152) compare les résultats de modélisation et commande, côté sortie, le trajet de transmission supplémentaire (148).
  38. Dispositif selon la revendication 37, caractérisé en ce que le trajet de transmission supplémentaire comprend des moyens de retard de temps pouvant être commandés, de préférence avec l'algorithme WSOLA.
  39. Prothèse auditive avec une unité de calcul qui modélise la perception d'au moins une grandeur psycho-acoustique par l'homme à des signaux acoustiques reçus,
    caractérisée en ce que l'unité de calcul paramètre la grandeur psycho-acoustique avec des bandes de fréquence critiques de l'ouïe humaine.
  40. Prothèse auditive selon la revendication 39,
    caractérisée en ce que l'unité de calcul calcule le modèle avec au moins deux jeux de paramètres, partant respectivement de signaux d'entrée et de sortie de la prothèse auditive et règle en fonction de la différence de modèle la transmission entre des signaux d'entrée et de sortie.
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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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
DK95103571T DK0661905T3 (da) 1995-03-13 1995-03-13 Fremgangsmåde til tilpasnning af et høreapparat, anordning hertil og høreapparat
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

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EP95103571A EP0661905B1 (fr) 1995-03-13 1995-03-13 Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive

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EP2278827A1 (fr) 2006-03-23 2011-01-26 Phonak Ag Procedé pour l'adaptation individuelle d'un appareil auditif
US8019095B2 (en) 2006-04-04 2011-09-13 Dolby Laboratories Licensing Corporation Loudness modification of multichannel audio signals
US8090120B2 (en) 2004-10-26 2012-01-03 Dolby Laboratories Licensing Corporation Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US8144881B2 (en) 2006-04-27 2012-03-27 Dolby Laboratories Licensing Corporation Audio gain control using specific-loudness-based auditory event detection
US8199933B2 (en) 2004-10-26 2012-06-12 Dolby Laboratories Licensing Corporation Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US8315398B2 (en) 2007-12-21 2012-11-20 Dts Llc System for adjusting perceived loudness of audio signals
US8396574B2 (en) 2007-07-13 2013-03-12 Dolby Laboratories Licensing Corporation Audio processing using auditory scene analysis and spectral skewness
US8437482B2 (en) 2003-05-28 2013-05-07 Dolby Laboratories Licensing Corporation Method, apparatus and computer program for calculating and adjusting the perceived loudness of an audio signal
US8504181B2 (en) 2006-04-04 2013-08-06 Dolby Laboratories Licensing Corporation Audio signal loudness measurement and modification in the MDCT domain
US8521314B2 (en) 2006-11-01 2013-08-27 Dolby Laboratories Licensing Corporation Hierarchical control path with constraints for audio dynamics processing
US8538042B2 (en) 2009-08-11 2013-09-17 Dts Llc System for increasing perceived loudness of speakers
US8849433B2 (en) 2006-10-20 2014-09-30 Dolby Laboratories Licensing Corporation Audio dynamics processing using a reset
US9312829B2 (en) 2012-04-12 2016-04-12 Dts Llc System for adjusting loudness of audio signals in real time

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JP4336457B2 (ja) 1999-08-17 2009-09-30 フォーナック アーゲー 補聴器調整装置
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US8437482B2 (en) 2003-05-28 2013-05-07 Dolby Laboratories Licensing Corporation Method, apparatus and computer program for calculating and adjusting the perceived loudness of an audio signal
US8090120B2 (en) 2004-10-26 2012-01-03 Dolby Laboratories Licensing Corporation Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US8199933B2 (en) 2004-10-26 2012-06-12 Dolby Laboratories Licensing Corporation Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US9350311B2 (en) 2004-10-26 2016-05-24 Dolby Laboratories Licensing Corporation Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
US8488809B2 (en) 2004-10-26 2013-07-16 Dolby Laboratories Licensing Corporation Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal
EP2278827A1 (fr) 2006-03-23 2011-01-26 Phonak Ag Procedé pour l'adaptation individuelle d'un appareil auditif
US7715571B2 (en) 2006-03-23 2010-05-11 Phonak Ag Method for individually fitting a hearing instrument
US8504181B2 (en) 2006-04-04 2013-08-06 Dolby Laboratories Licensing Corporation Audio signal loudness measurement and modification in the MDCT domain
US8019095B2 (en) 2006-04-04 2011-09-13 Dolby Laboratories Licensing Corporation Loudness modification of multichannel audio signals
US9584083B2 (en) 2006-04-04 2017-02-28 Dolby Laboratories Licensing Corporation Loudness modification of multichannel audio signals
US8600074B2 (en) 2006-04-04 2013-12-03 Dolby Laboratories Licensing Corporation Loudness modification of multichannel audio signals
US8428270B2 (en) 2006-04-27 2013-04-23 Dolby Laboratories Licensing Corporation Audio gain control using specific-loudness-based auditory event detection
US11962279B2 (en) 2006-04-27 2024-04-16 Dolby Laboratories Licensing Corporation Audio control using auditory event detection
US9136810B2 (en) 2006-04-27 2015-09-15 Dolby Laboratories Licensing Corporation Audio gain control using specific-loudness-based auditory event detection
US8144881B2 (en) 2006-04-27 2012-03-27 Dolby Laboratories Licensing Corporation Audio gain control using specific-loudness-based auditory event detection
US9450551B2 (en) 2006-04-27 2016-09-20 Dolby Laboratories Licensing Corporation Audio control using auditory event detection
US8849433B2 (en) 2006-10-20 2014-09-30 Dolby Laboratories Licensing Corporation Audio dynamics processing using a reset
US8521314B2 (en) 2006-11-01 2013-08-27 Dolby Laboratories Licensing Corporation Hierarchical control path with constraints for audio dynamics processing
US8396574B2 (en) 2007-07-13 2013-03-12 Dolby Laboratories Licensing Corporation Audio processing using auditory scene analysis and spectral skewness
US8315398B2 (en) 2007-12-21 2012-11-20 Dts Llc System for adjusting perceived loudness of audio signals
US9264836B2 (en) 2007-12-21 2016-02-16 Dts Llc System for adjusting perceived loudness of audio signals
US9820044B2 (en) 2009-08-11 2017-11-14 Dts Llc System for increasing perceived loudness of speakers
US8538042B2 (en) 2009-08-11 2013-09-17 Dts Llc System for increasing perceived loudness of speakers
US9312829B2 (en) 2012-04-12 2016-04-12 Dts Llc System for adjusting loudness of audio signals in real time
US9559656B2 (en) 2012-04-12 2017-01-31 Dts Llc System for adjusting loudness of audio signals in real time

Also Published As

Publication number Publication date
ATE229729T1 (de) 2002-12-15
EP1207718A3 (fr) 2003-02-05
DK0661905T3 (da) 2003-04-07
EP1207718A2 (fr) 2002-05-22
EP0661905A3 (fr) 1995-10-04
DE59510501D1 (de) 2003-01-23
EP0661905A2 (fr) 1995-07-05

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