EP0820212B1 - Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic - Google Patents

Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic Download PDF

Info

Publication number
EP0820212B1
EP0820212B1 EP97810460A EP97810460A EP0820212B1 EP 0820212 B1 EP0820212 B1 EP 0820212B1 EP 97810460 A EP97810460 A EP 97810460A EP 97810460 A EP97810460 A EP 97810460A EP 0820212 B1 EP0820212 B1 EP 0820212B1
Authority
EP
European Patent Office
Prior art keywords
filter
signal
value
interpolation
values
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.)
Expired - Lifetime
Application number
EP97810460A
Other languages
German (de)
English (en)
Other versions
EP0820212A3 (fr
EP0820212A2 (fr
Inventor
Arthur Schaub
Remo Leber
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.)
Bernafon AG
Original Assignee
Bernafon AG
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Bernafon AG filed Critical Bernafon AG
Publication of EP0820212A2 publication Critical patent/EP0820212A2/fr
Publication of EP0820212A3 publication Critical patent/EP0820212A3/fr
Application granted granted Critical
Publication of EP0820212B1 publication Critical patent/EP0820212B1/fr
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Images

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • 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
    • H04R25/35—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using translation techniques
    • H04R25/356—Amplitude, e.g. amplitude shift or compression

Definitions

  • the invention relates to a method for the loudness-controlled processing of acoustic signals in sound processing equipment and to an apparatus for carrying out the method according to the preambles of the independent claims.
  • the invention is particularly suitable for use in hearing aids for the hearing impaired; incoming acoustic signals are processed in such a way that the loudness subjectively perceived by the hearing impaired again always corresponds to the loudness perceived by normal hearing persons.
  • the idea of the loudness-controlled processing of acoustic signals has been known for some time and has been described by various authors in the specialist literature, for example also by N. Dillier et al. in the Journal of Rehabilitation Research and Development, Vol. 1, 1993, pp. 100-103 , The method is based on the fact that normal hearing and hearing impaired known test signals for assessing the subjectively perceived loudness are presented. As test signals, harmonic sine signals or narrow band noise are used. The subjectively perceived loudness depends on the signal power and on the frequency of a sine signal, or on the frequency of the dominant signal components of a complex signal. The subjective loudness data are on a normalized scale with value range [0, 1].
  • hearing loss-specific, loudness-dependent correction data By comparing the information of a hearing impaired person with those of a reference group of normal hearing impaired, hearing loss-specific, loudness-dependent correction data can be determined. In a suitable signal processing method, these correction data are then used to prepare the hearing impaired the acoustic signals of his environment in the targeted manner. In the aforementioned article, in comprehensibility tests with a group of 13 hearing impaired, remarkable improvements in intelligibility were demonstrated.
  • the loudness-controlled processing in the previously known form can not be used in practice. Namely, as described in the article mentioned, the processing takes place by Fourier transformation of short signal segments, modification of the short-term spectra and back transformation of the modified short-term spectra into the time domain. Due to the segment-by-segment processing, the processed signal has a delay of almost 20 ms. In intelligibility tests, this delay does not matter. However, in practice, when the hearing impaired person also speaks and perceives his own voice as being delayed, it is totally unacceptable. In the method described in the article mentioned the duration of the individual segments is 12.8 ms, and this value can not be significantly undershot, because to obtain a useful short-term spectrum, a minimum segment duration of this magnitude is essential.
  • the loudness model used in the processing In contrast to the simple test signals, the signal power of speech, music and noise is distributed over a wide frequency interval in a time-dependent and complex manner. With a loudness model, a loudness value is assigned to these complex signals in a time-dependent manner, which in the ideal case coincides exactly with the loudness perceived by normal hearing persons. The value determined with the loudness model is used for the time-dependent control of the signal processing.
  • the loudness model described in the article mentioned takes into account not only the total energy of a signal segment but also the center of gravity frequency of its short-term spectrum. To calculate the center of gravity frequency is based on fundamentals of E.
  • the loudness subjectively perceived by the hearing aid user should always correspond to the loudness perceived by normal hearing persons.
  • the signal delay should be so small that a hearing aid user is not irritated by the delayed perception of his own voice when speaking. It should also reduce the computational resources compared to known methods for the loudness-controlled processing of acoustic signals.
  • an apparatus for carrying out the method according to the invention is to be created.
  • the processing of the acoustic signal without Fourier transformation takes place, ie completely in the time domain, and also without division into subband signals.
  • the peculiarity of the inventive method is that a characteristic of the loudness control ⁇ is calculated in an iterative manner and used to control a time-dependent correction filter.
  • iterative calculation mode is meant that a new value is calculated for the control variable ⁇ at each sampling time, using values which have the quantities necessary for their calculation in the respectively preceding sampling time.
  • the loudness-specific control variable is thus determined not only as an average of successive signal segments but as a continuous time function.
  • the short signal delay typically measured at 2 ms, represents the observation time required for reliable estimation beyond the respective validity time point, and thus, unlike the segment-wise method, is not merely the result of an adverse property of the chosen implementation.
  • the iterative method of calculation takes place in the method according to the invention by means of particularly efficient and at the same time original method steps.
  • the time-dependent correction filter is controlled by assigning new values to parameters of the correction filter at each sampling time by interpolation with the aid of the control variable ⁇ .
  • the hearing-impaired correction data are stored as gain values for the individual spectral lines of a short-term spectrum
  • for well-defined values of the control variable ⁇ coefficient sets for prototype filters are determined in advance and stored. The transfer functions of these prototype filters run along the corresponding gain values, which are determined in the segment-by-segment procedure for the individual spectral lines of a short-term spectrum.
  • sets of coefficients are used in the process according to the invention which are known to be suitable for interpolation, ie that the transfer function determined by interpolated coefficients is expected to run between the transfer functions determined by the coefficient sets on which the interpolation is based.
  • the invention further relates to a device for carrying out the method according to the invention.
  • This device contains a stage for the iterative calculation of the loudness characteristic control variable ⁇ and thus a time-dependent controlled correction filter stage, which prepares incoming acoustic signals target setting.
  • the already mentioned drastic reduction of the required processing resources has different causes.
  • the iterative calculation method eliminates the segment-by-segment buffering of the input and output signals. Then, saving the coefficient sets for the prototype filters also results in a substantial saving compared to storing the gain values for the individual spectral lines of the short-term spectra.
  • FIG. 1 shows the use of the inventive method and the method itself in a schematic overview.
  • An acoustic signal is converted by a microphone 1 into an electrical signal, which is digitized by a signal converter 2 and then freed in a high-pass filter 3 of any offset and lowest-frequency interference signal components.
  • the essential steps of the method according to the invention consist in the processing of an output signal x of the high-pass filter 3.
  • a processing stage 4 the iterative calculation of the control variable ⁇ takes place.
  • the parameters of a time-dependent correction filter 7 are thus determined and transferred to this.
  • a delay stage 6 provides with respect to the filtering with the correction filter 7 for the synchronization of the signal x with the filter parameter values derived therefrom by causing a corresponding signal delay, for example by 2 ms.
  • the delay stage 6 is advantageously designed as a cyclic buffer with 32 memory locations.
  • the filtered with the correction filter 7 signal y passes to a signal converter 8 and is converted there into an analog electrical signal.
  • an analog amplifier stage 9 it is amplified with a hearing impaired but constant gain value g e , and then fed to an electro-acoustic signal converter 10.
  • the value of g e is determined in processing the coefficient sets for the prototype filters, such that the 16-bit number format used in the apparatus for performing the method is used as optimally as possible, with limitation of the processed signals due to the assumption in the apparatus However, saturation arithmetic only exceptionally effective.
  • the loudness of complex signals can be determined on the basis of the total energy of short signal segments and the center of gravity frequency of their short-term spectra.
  • the loudness depends approximately quadratically on the signal energy expressed on a logarithmic scale.
  • L represents the loudness limited to the value range [L min , L max ], and L min and L max are sensibly selected minimum and maximum values of loudness, thus defining the working range of the method within which the correction filter due to smallest changes the loudness is constantly updated.
  • the block diagram in Fig. 2 shows in more detail how the control variable ⁇ is obtained from the input signal x.
  • a momentary signal power q takes the place of the signal energy of a short signal segment and an instantaneous center of gravity c takes the place of the center frequency of its short-term spectrum.
  • These quantities are determined in processing stages 11-15.
  • corresponding output signal values c r and q r due to the iterative type of calculation still have an undesired scattering, which is eliminated in subsequent smoothing filters 14 and 15.
  • the smoothed signals c and q are applied in a processing stage 16 to the already mentioned two-dimensional interpolation, wherein the successive output signal values ⁇ r also have a still undesirable scattering, which is eliminated with a subsequent smoothing filter 17.
  • An essential aspect of the method according to the invention lies in the iterative calculation mode of the logarithmic signal power q as well as the center of gravity frequency c expressed on a bar scale, ie the conversion of the formula (1) into an iterative calculation scheme.
  • a frequency-selective weighting of the input signal x is made with a filter, which is hereinafter referred to as a frequency group filter.
  • the frequency group filter is in Fig. 2 shown as processing stage 11, and its output signal is denoted by ⁇ .
  • a frequency-selective weighting of the signal ⁇ is carried out in the method according to the invention with a filter which is referred to below as a bark filter.
  • the denominator in formula (4) in turn causes a normalization for the purpose of optimal use of the given number format.
  • the transfer function H B (f) is also approximated by a second order recursive digital filter 12, which in turn maps the in Fig. 3 having shown structure.
  • a simple first-order estimated value calculation unit is used for the time-exponentially weighted expected value of the squared input signal.
  • Such an estimated value calculation unit is for the general case, with input signal u and output signal v, in Fig. 4 shown.
  • a new output signal value v results from the fact that the output signal value of the preceding sampling instant is multiplied by the constant (1- ⁇ ) and the square of the new input signal value u multiplied by the constant factor ⁇ is added to this product.
  • the adaptation constant ⁇ for which 0 ⁇ ⁇ ⁇ 1 the speed at which the output signal v follows the changing input signal power can be controlled.
  • the simple estimate calculation unit of Fig. 4 has the disadvantages that a doubly wide number format is required for processing the squared input signal, and that the logarithm of the output signal v is additionally required for the subsequent calculations. Both Aspects in the inventive method in a simple way, as in Fig. 5 by embedding the simple estimate computing unit of Fig. 4 solved in a digital control loop.
  • the operation of the signal flow diagram in Fig. 5 is based on the fact that the variable v is controlled to a fixed setpoint. For this purpose, for each newly calculated signal value v, the incremental logarithmic increase or decrease in the signal power is determined, which corresponds to the deviation of the value v from the predetermined desired value.
  • the desired logarithmic signal power p results in the sequence by simply accumulating the successive incremental change values.
  • the iterative calculation of the center of gravity frequency is based on the calculation of the quotient of the signal powers of the signals ⁇ and ⁇ , for example in the processing stage 13.
  • the calculation of the signal powers is based on the in Fig. 5 mapped signal flow diagram returned. This results in the calculation of the center of gravity frequency in Fig. 8 illustrated signal flow diagram.
  • the lower part of the diagram is identical to the Fig. 5 , It is used to calculate the power of the signal ⁇ .
  • the upper part is used to calculate the power of the signal ⁇ .
  • the scaling and adjustment values are taken from the lower part of the circuit, which compares the signal flow diagram in the upper part Fig. 5 simplified. With this arrangement, the optimal use of the number format is also guaranteed for the calculation of the power of the signal ⁇ , and the sought center frequency results, as mentioned, by quotient of the two signal powers.
  • the loudness can be determined from the signal power p and the center of gravity frequency c.
  • the direct solution would be to place the signal flow diagrams in the Fig. 5 and 8th and their output signals after passing through smoothing filters of the interpolation stage 16 (see Fig. 2 ).
  • the method according to the invention involves a further substantial simplification due to the fact that the frequency group filter 11 performs only a frequency-selective weighting of the input signal x. This makes it possible to modify the entries in the original interpolation tables in such a way that the same value results for the control variable ⁇ when, instead of the logarithmic signal power p of the input signal x, the logarithmic signal power q of the signal ⁇ is used together with the modified tables. This eliminates the separate calculation of the signal power p, and the processing stage 13 in the inventive method Fig. 2 includes only the in Fig. 8 illustrated signal flow diagram.
  • a new output value c is obtained by adding a correction quantity D to the output value of the preceding sampling instant.
  • the correction quantity D is determined from the difference d, which results from the new input signal value c r and the previous output signal value.
  • the quantity d is first multiplied by a constant factor ⁇ > 1.
  • the value of ⁇ is set to, for example, 2 and 3, respectively, and the result of the multiplication is limited to the value range [-1, 1] by a saturation arithmetic.
  • the product w is then squared and limited to a value ⁇ , and the correction quantity D is obtained by multiplying the value thus calculated by the quantity w.
  • mapping curve D (d) is composed of five different curve parts 27.1-27.5.
  • the correction quantity D in the third power depends on the difference d; this corresponds to a first curve part 27.1.
  • mapping curve D (d) merges into linear parts; this corresponds to a second and third curve part 27.2 and 27.3. With significant changes in the input signal, these parts make sure that the output signal follows with only minimal delay.
  • fourth and fifth parts 27.4 and 27.5 of the imaging curve, where a limit is set to a constant value, guarantees a smooth transition even in the case of extremely unsteady changes in the input signal d.
  • the schema comprises three tables.
  • the table designated by ⁇ 0 contains the interpolation point values for fixed values of the input quantities c and q.
  • the other two tables, denoted by ⁇ / ⁇ c and ⁇ / ⁇ q, contain the gradient values of the function ⁇ (c, q) in the direction of the c and q coordinates, which match the interpolation points.
  • ⁇ r ⁇ 0 (c i , q k ) + (c - c i ) ⁇ ( ⁇ / ⁇ c)
  • c i and q k represent the nearest to c and q interpolation point coordinates, which are at the same time not greater than c or q itself.
  • Another aspect of the method according to the invention relates to the use of optimal table values in two-dimensional interpolation.
  • the Values of the function ⁇ (c, q) at the corners of a rectangle defined by successive vertex coordinates are shown schematically as ⁇ (c i , q k ), ⁇ (c i + 1 , q k ), ⁇ (c i , q k + 1 ) and ⁇ (c i + 1 , q k + 1 ).
  • the unavoidable interpolation errors are more evenly distributed than with the approximate table values ⁇ (c i , q k ), [ ⁇ (c i + 1 , q k ) - ⁇ (c i , q k )] and [ ⁇ (c i , q k + 1 ) - ⁇ (c i , q k )].
  • the successive signal values ⁇ r have an undesirable scattering, which is combined with the smoothing filter 17 (cf. Fig. 2 ) is eliminated.
  • the output signal of the smoothing filter 17 is the control variable ⁇ , which in the interpolation stage 5 (see. Fig. 1 ) is used to determine filter parameters of the correction filter 7.
  • the interpolation stage 5 is in the block diagram of Fig. 12 shown in more detail.
  • the control variable ⁇ arrives at a processing stage 18, from where it is masked out for the subsequent interpolations by the in Fig. 13 shown bit fields a table address ⁇ a and a proportional size ⁇ f are obtained.
  • a processing stage 19 represents a 3-bit-wide counter whose count is denoted by j.
  • a gain value g of the correction filter 7 is determined, and in a processing stage 21, filter coefficients k j (n) and k j (p) are determined.
  • the count value j and the interpolated filter parameters g, k j (n) and k j (p) are designated as a whole with m
  • the count value j and the interpolated filter parameters g, k j (n) and k j (p) reach the correction filter 7, which is shown in the block diagram of FIG Fig. 14 is shown in more detail. It comprises an amplifier stage 22, a cross-member filter 24 for the realization of zeros and a cross-member filter 26 for the realization of poles. For completeness, the structures of the cross-member filters 24 and 26 in the signal flow diagrams of FIGS. 15 or 16 reproduced in detail.
  • an interpolated gain value g reaches the amplifier stage 22 (cf. Fig. 14 ) and is multiplied by the input signal x d delayed by, for example, 2 ms.
  • the filter coefficients k j (n) and k j (p) reach processing stages 23 and 25, respectively, to which the counter value j is also passed.
  • the processing stages 23 and 25 are merely switches which assign the interpolated filter coefficient values corresponding to the counter value j to the correct filter coefficients in the cross-member filters 24 and 26, respectively.
  • the counter values 0 to 7 are assigned the filter coefficients with the indices 1 to 8 in ascending order.
  • the interpolation stages 20 and 21 are in the Figures 17 or 18 detailed.
  • the hearing correction data determined from the individual loudness data in the method according to the invention are stored as filter parameters in a form suitable for interpolation.
  • the table ⁇ is omitted and the corresponding value can be recalculated each time by subtracting the read value ⁇ 0 from the value tabulated below.
  • the Fig. 17 thus represents a two-stage interpolation scheme, which makes use of the normality of the signal values and matched tables for the efficient determination of the required output value.
  • the hearing-impaired-specific values are stored in the form of the log area-ratio coefficients.
  • the modulo 7 counter represented by the processing stage 19 controls the selection mechanism. In the two-stage interpolation scheme of Fig. 18 is therefore the three-bit value of the counter with the size ⁇ a joined to the current table address.
  • the filter coefficients k j (n) and k j (p) required in the cross-member filters 24 and 26 are determined in a renewed interpolation, wherein from each of the log area ratio coefficients ⁇ , in turn, by masking out the in Fig. 20 shown bit fields, an address value ⁇ a and a proportional variable ⁇ f are obtained.
  • this process, as well as the subsequent interpolation itself, can take place one after the other, which in Fig. 18 is indicated with the multiplexer M and in particular has the consequence that the tanh and ⁇ tanh designated tables of the tangent hyperbolic function must be stored only once.
  • an acoustic signal x to be processed is completely in the time domain is processed.
  • a control variable ⁇ which is characteristic of the subjective loudness sensation is continuously calculated.
  • the input signal x is processed with a time-dependent filter 7, the parameters of which are newly determined by means of the control variable ⁇ continuously by interpolation in user-specific correction data previously calculated and stored in tables and applied to the time-dependent filter 7.
  • a device according to the invention for carrying out the method has a processing stage 4 for the iterative calculation of the control variable ⁇ and a time-dependent controlled correction filter stage 7.

Landscapes

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

Claims (29)

  1. Procédé de traitement, commandé par la sonie, de signaux acoustiques dans des appareils de traitement du son, caractérisé en ce qu'on traite complètement dans le domaine temporel un signal acoustique (x) à traiter, procédé dans lequel, en partant du signal (x) à traiter, on calcule en continu une variable de référence (ψ), caractéristique de la sensation auditive subjective de personnes ayant une audition normale, et on traite le signal d'entrée (x) avec un filtre (7) dépendant du temps, on détermine de nouveau ses paramètres en continu, à l'aide de la variable de référence (ψ), par interpolation dans des données de correction calculées au préalable, mémorisées dans des tableaux et spécifiques de l'utilisateur, et on les utilise sur lé filtre (7) dépendant du temps.
  2. Procédé selon la revendication 1, caractérisé en ce que le signal acoustique (x) est traité d'une manière itérative sans répartition en signaux de sous-bandes.
  3. Procédé selon la revendication 1 ou 2, caractérisé en ce que la variable de référence (ψ) est définie comme étant la racine de la sonie normée sur un intervalle de sonie limité.
  4. Procédé selon l'une des revendications 1 à 3, caractérisé en ce que la variable de référence (ψ) est déterminée en continu par une interpolation bidimensionnelle, et plus précisément à l'aide de deux grandeurs calculées par itération, dont une première grandeur (p), calculée par itération, est une valeur estimée, exprimée sur une échelle logarithmique de la puissance instantanée du signal, et dont une deuxième grandeur (c), calculée par itération, est une valeur estimée, exprimée sur une échelle de Bark, de la fréquence moyenne de la distribution instantanée des puissances du signal.
  5. Procédé selon la revendication 4, caractérisé en ce que la première grandeur (p) calculée par itération est déterminée à l'aide d'une unité de calcul de la valeur estimée, incorporée dans une boucle de régulation numérique, travaillant par itération, du premier ordre pour une valeur escomptée, exponentiellement pondérée dans le temps, du signal d'entrée élevé au carré.
  6. Procédé selon la revendication 4 ou 5, caractérisé en ce que la deuxième grandeur (c) calculée par itération est calculée par division d'un dividende, déterminé par itération, par un diviseur déterminé par itération, le diviseur étant une valeur estimée de la puissance momentanée du signal (ψ) pondéré par un filtre de bande critique, et le dividende étant une valeur estimée de la puissance instantanée du signal (δ), qui en outre est encore pondéré par un filtre de Bark, la fonction de transfert du filtre de bande critique correspondant à la racine d'une fonction normée de largeur de bande critique, et celle du filtre de Bark correspondant à la racine d'une fonction normée de perceptions de la hauteur du son.
  7. Procédé selon la revendication 6, caractérisé en ce que tant le diviseur que le dividende sont déterminés à l'aide d'une unité de calcul de la valeur estimée, incorporée dans une boucle de régulation numérique, travaillant par itération, du premier ordre, pour une valeur escomptée, à pondération exponentielle dans le temps, du signal d'entrée élevé au carré, l'unité de détermination du dividende reçoit les signaux de régulation de celui du diviseur, et utilise ces signaux.
  8. Procédé selon la revendication 6, caractérisé en ce que la division est calculée à l'aide des grandeurs réglées de la valeur estimée, et est approximée par une multiplication (1 - δ), 1 représentant la valeur prescrite et |δ| << 1.
  9. Procédé selon l'une des revendications 5 à 8, caractérisé en ce que les grandeurs de mise à l'échelle nécessaires à la régulation de l'unité de calcul de la valeur estimée par itération, ainsi que les valeurs de modification incrémentielles, nécessaires à l'ajustement de la valeur estimée logarithmique, sont extraites de tableaux préalablement mémorisés (S, A, Δp).
  10. Procédé selon la revendication 9, caractérisé en ce que l'extraction à partir de tableaux organisés de cette manière a lieu de telle sorte que les indices de tableau peuvent, pour trouver les grandeurs recherchées, être obtenus par un simple masquage de champs de bits, à partir de la grandeur (v) de la valeur estimée non encore réglée, et de la grandeur logarithmique (p) de la valeur estimée.
  11. Procédé selon l'une des revendications 1 à 3, caractérisé en ce que la grandeur de référence (ψ) est déterminée en continu par une interpolation bidimensionnelle, et plus précisément à l'aide de deux grandeurs calculées par itération, dont une première grandeur (q) calculée par itération est une valeur estimée, exprimée sur une échelle logarithmique, de la puissance instantanée d'un signal (ψ) pondéré par un filtre de bande critique, la pondération étant compensée par une variation des entrées dans le tableau d'interpolation initial, et une deuxième grandeur (c), calculée par itération, est une valeur estimée, exprimée sur une échelle de Bark, de la fréquence moyenne de la distribution instantanée de la puissance du signal.
  12. Procédé selon l'une des revendications 4 à 11, caractérisé en ce que la valeur de référence (ψ), et/ou la première grandeur (p ou q) calculée par itération, et/ou la deuxième grandeur (c) calculée par itération, sont lissées par un filtre non alinéaire, plus précisément par le fait qu'une nouvelle valeur initiale s'obtient par addition d'une valeur de correction (D) à la valseur initiale précédente, cette valeur de correction (D) est calculée à partir de la différence (d) entre un nouveau signal d'entrée et le signal de sortie précédent, et la valeur de correction (D) dépend, pour de petites valeurs (|d|) de la différence (d), de la troisième puissance de cette différence (d) ; pour des valeurs moyennes (|d|) de la différence (d) dépend linéairement de cette différence (d) ; et pour de grandes valeurs (|d|) de la différence (d) est constante.
  13. Procédé selon l'une des revendications 4 à 12, caractérisé en ce que l'interpolation de la valeur de référence (ψ) a lieu à l'aide de tableaux organisés de telle sorte que tant l'indice de tableau, destiné à trouver la valeur d'appui et les grandeurs d'accroissement incrémentiel dans les deux dimensions, que les grandeurs proportionnelles, par lesquelles on multiplie les grandeurs d'accroissement incrémentiel avant addition à la valeur d'appui, peuvent être obtenues par simple marquage de champs de bits, à partir des grandeurs (p ou q; c) calculées par itération.
  14. Procédé selon la revendication 13, caractérisé en ce que, dans les tableaux, on utilise pour l'interpolation dans les deux dimensions de la valeur de référence (ψ) des valeurs optimisées selon les formules ψ 0 c i q k = ψ c i q k + ψ ⁢ c i + 1 q k + ψ ⁢ c i q k + 1 - ψ ⁢ c i + 1 q k + 1 - ψ c i q k / 4
    Figure imgb0009
     ∂ ψ / ∂ c | ci , qk = ψ ⁢ c i + 1 q k + 1 - ψ ⁢ c i q k + 1 + ψ ⁢ c i + 1 q k - ψ c i q k / 2
    Figure imgb0010
     et ∂ ψ / ∂ q | ci , qk = ψ ⁢ c i + 1 q k + 1 - ψ ⁢ c i + 1 q k + ψ ⁢ c i q k + 1 - ψ c i q k / 2
    Figure imgb0011
  15. Procédé selon l'une des revendications 1 à 14, caractérisé en ce que les valeurs stockées dans le tableau pour l'interpolation des données de correction spécifiques de l'utilisateur sont, en tant que valeurs d'amplification, déposées dans le domaine logarithmique et, en tant que coefficients de filtrage, dans le domaine Log-Area-Ratio.
  16. Procédé selon la revendication 15, caractérisé en ce que l'interpolation des données de correction spécifiques de l'utilisateur est réalisée avec des tableaux organisés de telle sorte que l'indice de tableau servant à trouver la valeur d'appui, ainsi que l'indice de tableau servant à trouver la grandeur proportionnelle par laquelle, ayant addition à la valeur d'appui, on multiplie la différence entre la valeur d'appui suivante et la valeur d'appui proprement dite, soient obtenus par un masquage simple de champs de bits à partir de la valeur de référence (ψ).
  17. Procédé selon la revendication 15 ou 16, caractérisé en ce que la valeur d'amplification est déterminée à partir de la valeur d'amplification logarithmique interpolée, et les coefficients de filtrage sont déterminés à partir des coefficients Log-Area-Ratio interpolés, une fois de plus par interpolation avec des tableaux mémorisés de la fonction exponentielle et de la fonction tangente hyperbolique, ainsi qu'avec des tableaux des grandeurs d'accroissement incrémentiel de ces fonctions.
  18. Procédé selon la revendication 17, caractérisé en ce que l'interpolation est réalisée avec des tableaux organisés de telle sorte que les indices de tableau servant à trouver les valeurs d'appui et les grandeurs d'accroissement incrémentiel, ainsi que les grandeurs proportionnelles par lesquelles, avant addition aux valeurs d'appui, on multiplie les grandeurs d'accroissement incrémentiel, soient obtenus par simple masquage de champs de bits de la valeur d'amplification interpolée et des coefficients Log-Area-Ratio interpolés.
  19. Procédé selon l'une des revendications 15 à 18, caractérisé en ce que la valeur d'amplification est, dans chaque intervalle de balayage, et, parmi les coefficients de filtrage dans chaque intervalle de balayage, seulement les coefficients d'une paire pôle-zéro sont déterminés de nouveau, une séquence uniforme et fixe servant au renouvellement des coefficients de filtrage.
  20. Procédé selon l'une des revendications 1 à 19, caractérisé en ce que le signal d'entrée allant vers le filtre mentionné ci-dessus dépendant du temps est retardé de telle sorte que les coefficients de filtrage et les valeurs d'amplification qui sont en permanence déterminés de nouveau par le calcul de la grandeur (ψ) mentionnée ci-dessus, soient utilisés au moment voulu sur le signal servant au calcul.
  21. Dispositif pour la mise en oeuvre du procédé selon la revendication 1, caractérisé par une étape de traitement (4) servant au calcul par itération de la valeur de référence (ψ), et par un étage de filtrage de correction (7), commandé en fonction du temps.
  22. Dispositif selon la revendication 21, caractérisé par un étage d'interpolation bidimensionnel (16) servant à déterminer la valeur de référence (ψ) à partir d'une puissance de signal (q) et d'une fréquence moyenne (c).
  23. Dispositif selon la revendication 21 ou 22, caractérisé par un filtre de bande critique (11) et un filtre de Bark (12) pour déterminer des signaux filtrés (ψ, δ) à partir d'un signal d'entrée (x).
  24. Dispositif selon la revendication 23, caractérisé en ce que le filtre de bande critique et le filtre de Bark mentionné ci-dessus sont configurés comme des filtres récursifs.
  25. Dispositif selon l'une des revendications 21 à 24, caractérisé par une unité (13) de calcul de la valeur estimée, destinée au calcul de la puissance du signal (q) et de la fréquence moyenne (c) à partir des signaux d'entrée filtrés (ψ, δ).
  26. Dispositif selon l'une des revendications 21 à 25, caractérisé par des filtres de lissage (14, 15, 17) destinés à éliminer une dispersion indésirable de valeurs successives du signal (cr, qr, (ψ)r).
  27. Dispositif selon l'une des revendications 21 à 26, caractérisé par un montage en série d'un étage d'amplification (22), d'un étage de filtrage (24) à montage en croix pour réaliser des zéros, et d'un étage de filtrage à montage en croix (26) pour la réalisation de pôles.
  28. Dispositif selon l'une des revendications 21 à 27, caractérisé par des étages d'interpolation en deux étages pour déterminer la valeur d'amplification (g) et les coefficients (kj (n) et kj (p)) du filtre de correction (7) à partir de la valeur de référence (ψ).
  29. Dispositif selon l'une des revendications 21 à 28, caractérisé par un signal de temporisation de signal (6), pour la synchronisation du signal d'entrée (x) pour ce qui concerne le traitement avec un filtre de correction (7) dont les paramètres de filtrage dérivent du signal d'entrée (x).
EP97810460A 1996-07-19 1997-07-11 Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic Expired - Lifetime EP0820212B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CH1823/96 1996-07-19
CH182396 1996-07-19

Publications (3)

Publication Number Publication Date
EP0820212A2 EP0820212A2 (fr) 1998-01-21
EP0820212A3 EP0820212A3 (fr) 2006-03-22
EP0820212B1 true EP0820212B1 (fr) 2010-04-21

Family

ID=4219434

Family Applications (1)

Application Number Title Priority Date Filing Date
EP97810460A Expired - Lifetime EP0820212B1 (fr) 1996-07-19 1997-07-11 Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic

Country Status (5)

Country Link
US (1) US6370255B1 (fr)
EP (1) EP0820212B1 (fr)
AU (1) AU729074B2 (fr)
DE (1) DE59713033D1 (fr)
DK (1) DK0820212T3 (fr)

Families Citing this family (26)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0820212B1 (fr) * 1996-07-19 2010-04-21 Bernafon AG Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic
US7277554B2 (en) * 2001-08-08 2007-10-02 Gn Resound North America Corporation Dynamic range compression using digital frequency warping
US7072477B1 (en) * 2002-07-09 2006-07-04 Apple Computer, Inc. Method and apparatus for automatically normalizing a perceived volume level in a digitally encoded file
US7454331B2 (en) * 2002-08-30 2008-11-18 Dolby Laboratories Licensing Corporation Controlling loudness of speech in signals that contain speech and other types of audio material
DE10245567B3 (de) * 2002-09-30 2004-04-01 Siemens Audiologische Technik Gmbh Vorrichtung und Verfahren zum Anpassen eines Hörgeräts
ES2290764T3 (es) * 2003-05-28 2008-02-16 Dolby Laboratories Licensing Corporation Metodo, aparato y programa de ordenador para calcular y ajustar la sonoridad percibida de una señal de audio.
WO2006047600A1 (fr) 2004-10-26 2006-05-04 Dolby Laboratories Licensing Corporation Calcul et reglage de la force sonore perçue et/ou de l'equilibre spectral perçu d'un signal audio
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
EA026063B1 (ru) * 2005-04-18 2017-02-28 Басф Се Сополимер, синтезированный из по меньшей мере трех различных моноэтиленненасыщенных мономеров
CN101410892B (zh) * 2006-04-04 2012-08-08 杜比实验室特许公司 改进的离散余弦变换域中的音频信号响度测量及修改
TWI517562B (zh) 2006-04-04 2016-01-11 杜比實驗室特許公司 用於將多聲道音訊信號之全面感知響度縮放一期望量的方法、裝置及電腦程式
CN102684628B (zh) 2006-04-27 2014-11-26 杜比实验室特许公司 修正音频动态处理器的参数的方法以及执行该方法的设备
KR101137715B1 (ko) 2006-10-20 2012-04-25 돌비 레버러토리즈 라이쎈싱 코오포레이션 리셋을 사용하는 오디오 다이내믹스 프로세싱
US8521314B2 (en) * 2006-11-01 2013-08-27 Dolby Laboratories Licensing Corporation Hierarchical control path with constraints for audio dynamics processing
JP4938862B2 (ja) * 2007-01-03 2012-05-23 ドルビー ラボラトリーズ ライセンシング コーポレイション ハイブリッドデジタル／アナログラウドネス補償音量調節
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
US8538042B2 (en) 2009-08-11 2013-09-17 Dts Llc System for increasing perceived loudness of speakers
EP2326108B1 (fr) * 2009-11-02 2015-06-03 Harman Becker Automotive Systems GmbH Égalisation de phase de système audio
NL2004294C2 (en) * 2010-02-24 2011-08-25 Ru Jacob Alexander De Hearing instrument.
US9313589B2 (en) * 2011-07-01 2016-04-12 Cochlear Limited Method and system for configuration of a medical device that stimulates a human physiological system
US9312829B2 (en) * 2012-04-12 2016-04-12 Dts Llc System for adjusting loudness of audio signals in real time
US9380387B2 (en) 2014-08-01 2016-06-28 Klipsch Group, Inc. Phase independent surround speaker
US10842418B2 (en) 2014-09-29 2020-11-24 Starkey Laboratories, Inc. Method and apparatus for tinnitus evaluation with test sound automatically adjusted for loudness
FR3052951B1 (fr) * 2016-06-20 2020-02-28 Arkamys Procede et systeme pour l'optimisation du rendu sonore de basses frequences d'un signal audio
EP3503574B1 (fr) * 2017-12-22 2021-10-27 FalCom A/S Dispositif de protection auditive doté d'un limitateur multibande et procédé associé

Family Cites Families (11)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5225836A (en) * 1988-03-23 1993-07-06 Central Institute For The Deaf Electronic filters, repeated signal charge conversion apparatus, hearing aids and methods
US5495534A (en) * 1990-01-19 1996-02-27 Sony Corporation Audio signal reproducing apparatus
US5388185A (en) * 1991-09-30 1995-02-07 U S West Advanced Technologies, Inc. System for adaptive processing of telephone voice signals
DE4340817A1 (de) * 1993-12-01 1995-06-08 Toepholm & Westermann Schaltungsanordnung für die automatische Regelung von Hörhilfsgeräten
EP0674463A1 (fr) * 1994-03-23 1995-09-27 Siemens Audiologische Technik GmbH Prothèse auditive programmable
US5500902A (en) * 1994-07-08 1996-03-19 Stockham, Jr.; Thomas G. Hearing aid device incorporating signal processing techniques
ATE229729T1 (de) * 1995-03-13 2002-12-15 Phonak Ag Verfahren zur anpassung eines hörgerätes, vorrichtung hierzu und hörgerät
JP2970498B2 (ja) * 1995-10-26 1999-11-02 日本電気株式会社 ディジタル補聴器
US5771299A (en) * 1996-06-20 1998-06-23 Audiologic, Inc. Spectral transposition of a digital audio signal
EP0820212B1 (fr) * 1996-07-19 2010-04-21 Bernafon AG Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic
IT1287089B1 (it) * 1996-11-07 1998-08-04 Curti Roberto Delle Dispositivo di equalizzazione/filtraggio per la correzione della linearita' di risposta dei sistemi di riproduzione e/o amplificazione

Also Published As

Publication number Publication date
EP0820212A3 (fr) 2006-03-22
US6370255B1 (en) 2002-04-09
DE59713033D1 (de) 2010-06-02
DK0820212T3 (da) 2010-08-02
AU2865597A (en) 1998-01-29
AU729074B2 (en) 2001-01-25
EP0820212A2 (fr) 1998-01-21

Similar Documents

Publication Publication Date Title
EP0820212B1 (fr) Traitement d&#39;un signal acoustique basé sur le contrôle de l&#39;intensité sonic
DE602004008455T2 (de) Verfahren, vorrichtung und computerprogramm zur berechung und einstellung der wahrgenommenen lautstärke eines audiosignals
EP0732036B1 (fr) Circuiterie de regulation automatique d&#39;appareils de correction auditive
DE69433662T2 (de) Adaptive verstärkung und filterschaltung für tonwiedergabesystem
DE69906560T2 (de) Cochlea-kompression modellbasiertes hörhilfegerät
EP1453194B1 (fr) Méthode de l&#39;ajustement automatique d&#39;un amplificateur dans une prothèse auditive et prothèse auditive
EP0052847B1 (fr) Procédé et circuit pour la conversion de la fréquence d&#39;échantillonnage d&#39;une suite d&#39;échantillons en évitant la conversion en un signal continu
DE2818204C2 (de) Signalverarbeitungsanlage zur Ableitung eines störverringerten Ausgangssignals
DE3510660C2 (fr)
DE60037167T2 (de) Steuerverfahren für einen digitalen graphischen Entzerrer
EP0661905B1 (fr) Procédé d&#39;adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive
DE69933141T2 (de) Tonprozessor zur adaptiven dynamikbereichsverbesserung
DE10234130B3 (de) Vorrichtung und Verfahren zum Erzeugen einer komplexen Spektraldarstellung eines zeitdiskreten Signals
EP0542711A1 (fr) Procédé pour le décalage en fréquence de signaux
DE102012220620A1 (de) Bereitstellen von Audiosignalen für eine Tinnitustherapie
EP1239455A2 (fr) Méthode et dispositif pour la réalisation d&#39;une transformation de Fourier adaptée à la fonction de transfert des organes sensoriels humains, et dispositifs pour la réduction de bruit et la reconnaissance de parole basés sur ces principes
DE602004006912T2 (de) Verfahren zur Verarbeitung eines akustischen Signals und ein Hörgerät
DE3504890A1 (de) Verfahren und vorrichtung zur frequenzmessung durch zaehlung der anzahl von nulldurchgaengen
EP0535425B1 (fr) Procédé d&#39;amplification de signaux acoustiques pour les malentendants et dispositif pour la réalisation du procédé
DE60125072T2 (de) Digitaler graphischer/parametrischer Entzerrer
DE4012349C2 (fr)
DE102012202469B3 (de) Hörvorrichtung mit einem adaptiven Filter und Verfahren zum Filtern eines Audiosignal
EP1453355A1 (fr) Traitement de signal dans un appareil auditif
CH655213A5 (de) Sprachverarbeitungsgeraet fuer teilnehmerleitungen.
DE60028769T2 (de) Anti-aliasierte begrenzung bei begrenzter modulation mit stufen-funktion

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

AK Designated contracting states

Kind code of ref document: A2

Designated state(s): AT BE CH DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE

PUAL Search report despatched

Free format text: ORIGINAL CODE: 0009013

AK Designated contracting states

Kind code of ref document: A3

Designated state(s): AT BE CH DE DK ES FI FR GB GR IE IT LI LU MC NL PT SE

17P Request for examination filed

Effective date: 20060920

AKX Designation fees paid

Designated state(s): CH DE DK FR GB LI

GRAP Despatch of communication of intention to grant a patent

Free format text: ORIGINAL CODE: EPIDOSNIGR1

GRAC Information related to communication of intention to grant a patent modified

Free format text: ORIGINAL CODE: EPIDOSCIGR1

RTI1 Title (correction)

Free format text: ACOUSTIC SIGNAL PROCESSING BASED ON LOUDNESS CONTROL

GRAS Grant fee paid

Free format text: ORIGINAL CODE: EPIDOSNIGR3

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): CH DE DK FR GB LI

REG Reference to a national code

Ref country code: GB

Ref legal event code: FG4D

Free format text: NOT ENGLISH

REG Reference to a national code

Ref country code: CH

Ref legal event code: EP

REF Corresponds to:

Ref document number: 59713033

Country of ref document: DE

Date of ref document: 20100602

Kind code of ref document: P

REG Reference to a national code

Ref country code: CH

Ref legal event code: NV

Representative=s name: FIAMMENGHI-FIAMMENGHI

REG Reference to a national code

Ref country code: DK

Ref legal event code: T3

PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

26N No opposition filed

Effective date: 20110124

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: CH

Payment date: 20130716

Year of fee payment: 17

Ref country code: DK

Payment date: 20130718

Year of fee payment: 17

Ref country code: DE

Payment date: 20130724

Year of fee payment: 17

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 20130731

Year of fee payment: 17

Ref country code: GB

Payment date: 20130725

Year of fee payment: 17

REG Reference to a national code

Ref country code: DE

Ref legal event code: R119

Ref document number: 59713033

Country of ref document: DE

REG Reference to a national code

Ref country code: DK

Ref legal event code: EBP

Effective date: 20140731

REG Reference to a national code

Ref country code: CH

Ref legal event code: PL

GBPC Gb: european patent ceased through non-payment of renewal fee

Effective date: 20140711

REG Reference to a national code

Ref country code: FR

Ref legal event code: ST

Effective date: 20150331

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: CH

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140731

Ref country code: LI

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140731

Ref country code: DE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20150203

REG Reference to a national code

Ref country code: DE

Ref legal event code: R119

Ref document number: 59713033

Country of ref document: DE

Effective date: 20150203

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: FR

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140731

Ref country code: GB

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140711

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DK

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20140731