EP0820212A2 - Traitement d'un signal acoustique basé sur le contrÔle de l'intensité sonore - Google Patents

Traitement d'un signal acoustique basé sur le contrÔle de l'intensité sonore Download PDF

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Publication number
EP0820212A2
EP0820212A2 EP97810460A EP97810460A EP0820212A2 EP 0820212 A2 EP0820212 A2 EP 0820212A2 EP 97810460 A EP97810460 A EP 97810460A EP 97810460 A EP97810460 A EP 97810460A EP 0820212 A2 EP0820212 A2 EP 0820212A2
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Prior art keywords
filter
signal
value
values
interpolation
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Granted
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EP97810460A
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German (de)
English (en)
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EP0820212B1 (fr
EP0820212A3 (fr
Inventor
Arthur Schaub
Remo Leber
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Bernafon AG
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Bernafon AG
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Electric hearing aids
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; 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; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Electric hearing aids
    • H04R25/35—Electric hearing aids using translation techniques
    • H04R25/356—Amplitude, e.g. amplitude shift or compression

Definitions

  • the invention relates to a method for loudness-controlled processing of acoustic signals in sound processing devices and a device 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 perceived by the hearing impaired always corresponds to the loudness perceived by normal hearing.
  • hearing impairment-specific, loudness-dependent correction data can be determined.
  • these correction data are then used to prepare the acoustic signals of the environment for the hearing impaired in the manner intended.
  • remarkable improvements in intelligibility were demonstrated in intelligibility tests with a group of 13 hearing impaired people.
  • processing is carried out by Fourier transforming short signal segments, modifying the short-term spectra and transforming the modified short-term spectra back into the time domain.
  • segmental processing there is a delay of almost 20 ms for the processed signal. This delay plays no role in intelligibility tests.
  • the hearing impaired person also speaks and perceives his own voice with such a delay, it is completely unacceptable.
  • the duration of the individual segments is 12.8 ms, and this value cannot be fallen short of significantly, because a minimum segment duration of this magnitude is essential in order to obtain a usable short-term spectrum.
  • loudness model used in the processing.
  • the signal power of speech, music and noises is distributed over a wide frequency range in a time-dependent and complex manner.
  • a loudness model a time-dependent loudness value is assigned to these complex signals, which ideally coincides exactly with the loudness felt by normal listeners.
  • 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. For the calculation of the center of gravity frequency, the basics of E.
  • the loudness subjectively felt by the hearing aid user should always correspond to the loudness felt 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.
  • Computational resources are also to be reduced compared to known methods for loudness-controlled processing of acoustic signals.
  • a device for carrying out the method according to the invention is to be created.
  • the acoustic signal is processed without Fourier transformation, that is to say completely in the time domain, and also without division into subband signals.
  • the special feature of the method according to the invention is that a control variable ⁇ characteristic of the loudness is calculated iteratively and used to control a time-dependent correction filter.
  • the expression "iterative calculation method” means that a new value is calculated for the control variable ⁇ at each sampling time, using values which had the variables necessary for their calculation at the previous sampling time.
  • the loudness-specific control variable is thus not only determined as the mean of successive signal segments, but rather as a continuous time function.
  • the short signal delay typically measured at 2 ms, represents the observation period required for reliable estimate formation beyond the respective point in time of validity and, in contrast to the segment-wise method, is therefore not merely the result of a disadvantageous property of the chosen implementation.
  • the iterative calculation is carried out 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 in that parameters of the correction filter are assigned new values at every sampling time by interpolation with the aid of the control variable ⁇ .
  • coefficient sets for prototype filters are determined and stored in advance for prototype filters. The transfer functions of these prototype filters run along the corresponding gain values, which are determined for the individual spectral lines of a short-term spectrum in the segment-wise method.
  • coefficient sets are used in the method according to the invention, from which are known to be suitable for interpolation, ie that the transfer function determined by interpolated coefficients runs as expected between the transfer functions which are determined by the sets of coefficients on which the interpolation is based.
  • the method according to the invention therefore breaks completely new ground.
  • the in the mentioned article by N. Dillier et al. achieved good understandability results.
  • the method according to the invention reduces the signal delay to approximately 2 ms and at the same time achieves a drastic reduction in the computational resources. It is therefore possible to implement the method according to the invention in a hearing aid of a conventional design today.
  • the invention further relates to a device for carrying out the method according to the invention.
  • This device contains a stage for iteratively calculating the control variable ⁇ which is characteristic of the loudness, and a correction filter stage which is thus controlled in a time-dependent manner and processes incoming acoustic signals in accordance with the objectives.
  • the aforementioned drastic reduction in processing resources has various causes.
  • the iterative calculation method eliminates the segmental buffering of the input and output signals. Then, when saving the coefficient sets for the prototype filter, there is also a substantial saving compared to saving the gain values for the individual spectral lines of the short-term spectra.
  • FIG. 1 shows the use of the method according to the invention 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 of any offset and extremely low-frequency interference signal components in a high-pass filter 3.
  • 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.
  • the processing variable 4 is used for the iterative calculation of the control variable ⁇ .
  • the parameters of a time-dependent correction filter 7 are thus determined and transferred to it.
  • a delay stage 6 provides for the filtering with the correction filter 7 the synchronization of the signal x with the filter parameter values derived from it 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 signal y filtered with the correction filter 7 arrives at a signal converter 8 and is converted there into an analog electrical signal.
  • an analog amplifier stage 9 it is amplified with a gain value g e specific to the hearing impaired but constant over time and then fed to an electro-acoustic signal converter 10.
  • the value of g e is determined during the preparation of the coefficient sets for the prototype filters, in such a way that the 16 bit wide number format used in the device for carrying out the method is used as optimally as possible, with a limitation of the processed signals as a result of the presupposed in the device Saturation arithmetic should only be effective in exceptional cases.
  • 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 roughly 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 chosen minimum and maximum values of loudness, which thus define the working range of the method within which the correction filter due to the smallest changes the loudness is constantly tracked.
  • the block diagram in FIG. 2 shows in somewhat more detail how the control variable ⁇ is obtained from the input signal x.
  • the instantaneous signal power q takes the place of the signal energy of a short signal segment and the instantaneous center of gravity frequency c replaces the center frequency of its short-term spectrum.
  • These sizes are determined in processing stages 11-15.
  • corresponding output signal values c r and q r still have an undesired scatter due to the iterative type of calculation, which is eliminated in subsequent smoothing filters 14 and 15.
  • the smoothed signals c and q are fed in a processing stage 16 to the two-dimensional interpolation already mentioned, the successive output signal values ⁇ r also having an undesirable scatter, which is eliminated with a subsequent smoothing filter 17.
  • An essential aspect of the method according to the invention lies in the iterative calculation type of the logarithmic signal power q and the center of gravity frequency c expressed on a Bark scale, that is to say the conversion of the formula (1) into an iterative calculation scheme.
  • frequency-selective weighting of the input signal x is carried out with a filter, which is referred to below as a frequency group filter.
  • the frequency group filter is shown in Fig. 2 as a processing stage 11, and its output signal is denoted by ⁇ .
  • a frequency-selective weighting of the signal ⁇ is carried out with a filter, which is also referred to as a Bark filter.
  • the denominator in formula (4) brings about standardization 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 has the structure shown in FIG. 3 .
  • a simple first-order estimate calculation unit for the exponentially weighted expected value of the squared input signal is used in the method according to the invention.
  • Such an estimated value calculation unit is shown in FIG. 4 for the general case, with input signal u and output signal v.
  • a new output signal value v results from the fact that the output signal value of the previous sampling time 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 applies, the speed at which the output signal v follows the changing input signal power can be controlled.
  • the functioning of the signal flow diagram in FIG. 5 is based on the fact that the variable v is regulated to a fixed predetermined setpoint.
  • the incremental logarithmic increase or decrease in the signal power is determined for each newly calculated signal value v, which corresponds to the deviation of the value v from the predetermined setpoint.
  • the logarithmic signal power p sought results subsequently from merely accumulating the successive incremental change values.
  • each input signal value x is scaled with a scaling factor that corresponds to the estimated value p, and that the variable v itself is also multiplied by an adjustment value corresponding to the change in power before it is updated again.
  • both the incremental change and the scaling and adjustment values are determined in the method according to the invention at each sampling time for values of the variables v and p, the accuracy of which is limited by cutting to 6 or 7 decimal places.
  • This enables the efficient use of tables in which the 64 or 128 previously calculated suitable values are stored.
  • the relevant bit fields need only be extracted from the variables v and p, as shown in FIGS. 6 and 7 .
  • the table with the incremental logarithmic power changes is designated by ⁇ p.
  • table S in FIG. 5 also contains modified scaling values that were obtained from the original scaling values by multiplying by the root from the constant ⁇ .
  • the adjustment values in the table labeled A have already been multiplied by the constant (1 - ⁇ ).
  • the usual 16-bit wide fixed-point number format is sufficient for storing the variables v and p and all the table values in FIG. 5.
  • 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 processing stage 13.
  • the calculation of the signal powers is traced back to the signal flow diagram shown in FIG. 5.
  • the lower part of the diagram is identical to 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 circuit part, which simplifies the signal flow diagram in the upper part compared to FIG. 5.
  • the optimal use of the number format is also guaranteed for the calculation of the power of the signal ⁇ , and the desired center of gravity results, as mentioned, by forming the quotient of the two signal powers.
  • the quotient Q Z / N formed from a numerator Z and a denominator N is calculated on the basis of the signal power values already tracked with an adjustment value from Table A.
  • the denominator takes on a numerical format standardized to the specified target value only slightly different from 1, and instead of dividing by (1 + ⁇ ), the quotient Q ⁇ Z (1 - ⁇ ) (7) by multiplying the counter Z by (1 - ⁇ ).
  • the loudness can be determined from the signal power p and the center of gravity frequency c.
  • the direct solution would be to use the signal flow diagrams in FIGS. 5 and 8 and to feed their output signals to the interpolation stage 16 (see FIG. 2) after passing through suitable smoothing filters.
  • the method according to the invention includes a further significant simplification due to the fact that the frequency group filter 11 only carries out a frequency-selective weighting of the input signal x. This makes it possible to modify the entries in the original interpolation tables so that the same value results for the control variable ⁇ if, 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.
  • the separate calculation of the signal power p is thus omitted in the method according to the invention, and the processing stage 13 in FIG. 2 only includes the signal flow diagram shown in FIG. 8.
  • a new output value c results from adding a correction quantity D to the output value of the previous sampling time.
  • 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 2 or 3, for example, and the result of the multiplication is limited to the value range [-1, 1] using 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 calculated in this way by the quantity w.
  • FIG. 10 shows the relationship between the internal variables d and D.
  • these smoothing filters make use of the normalization of the signals to be filtered, that is to say that their value range comprises the interval [0, 1].
  • the difference d thus takes on values from the interval [-1, 1].
  • the mapping curve D (d) shown in FIG. 10 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 part of the curve 27.1.
  • mapping curve D (d) changes into linear parts; this corresponds to a second and third curve part 27.2 and 27.3. In the event of significant changes in the input signal, these parts ensure that the output signal with minimal delay follows.
  • the control variable ⁇ is calculated in processing stage 16 with the filtered center of gravity frequency c and the filtered signal power q. As already mentioned, this process takes place by means of a two-dimensional interpolation, which is shown in FIG. 11 in a detailed scheme.
  • the scheme comprises three tables.
  • the table labeled ⁇ 0 contains the base point values for fixed values of the input variables c and q.
  • the other two tables, labeled ⁇ / ⁇ c and ⁇ / ⁇ q, contain the gradient values of the function ⁇ (c, q) that match the reference points in the direction of the c and q coordinates.
  • ⁇ r ⁇ 0 (c i , q k ) + (c - c i ) ⁇ ( ⁇ / ⁇ c)
  • ci, qk (8)
  • c i and q k represent the base coordinates closest to c and q, which are at the same time no larger than c or q itself.
  • the values c i and q k and (c-c i ) and (q-q k ) can be simply masked out in FIG. 11 in the method according to the invention Determine bit fields from sizes c and q. Finally, the values c i and q k combined according to FIG. 11 are used to address the table values.
  • 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 reference point coordinates are schematically represented by ⁇ (c i , q k ), ⁇ (c i + 1 , q k ), ⁇ (c i , q k + 1 ) and ⁇ (c i + 1 , q k + 1 ).
  • Interpolation stage 5 is shown in more detail in the block diagram of FIG. 12 .
  • the control variable ⁇ reaches a processing stage 18, from which a table address ⁇ a and a proportional variable gr f are obtained for the subsequent interpolations by masking out the bit fields shown in FIG. 13 .
  • a processing stage 19 represents a 3-bit wide counter, the count of which is denoted by j.
  • a gain value g of the correction filter 7 is determined in a processing stage 20 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 by m
  • the count value j and the interpolated filter parameters g, k j (n) and k j (p) arrive at the correction filter 7, which is shown in more detail in the block diagram in FIG. 14 .
  • It comprises an amplifier stage 22, a cross-section filter 24 for realizing zero points and a cross-section filter 26 for realizing pole positions.
  • the structures of the cross-link filters 24 and 26 are shown in detail in the signal flow diagrams in FIGS. 15 and 16, respectively .
  • an interpolated gain value g reaches amplifier stage 22 (see 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) arrive at processing stages 23 and 25, to which the counter value j is also led.
  • 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 coefficient in the cross-link filters 24 and 26, respectively.
  • the filter values with the indices 1 to 8 are assigned to the counter values 0 to 7 in ascending order.
  • the interpolation stages 20 and 21 are shown in detail in FIGS. 17 and 18 .
  • the hearing correction data determined from the individual loudness data are stored in the method according to the invention as filter parameters in a form suitable for interpolation.
  • the table ⁇ is omitted and the corresponding value can be recalculated each time by forming the difference between the read value ⁇ 0 and the value tabulated below.
  • FIG. 17 thus represents a two-stage interpolation scheme, which in turn makes use of the normalization of the signal values and tables matched to them for the efficient determination of the required output value.
  • the hearing-specific values are stored in the form of the log area ratio coefficients.
  • the modulo 7 counter represented by processing stage 19 controls the selection mechanism.
  • the three-bit value of the counter is therefore the size ⁇ a combined to the current table address.
  • the filter coefficients k j (n) and k j (p) required in the cross-link filters 24 and 26 are determined in a renewed interpolation, with each of the log area ratio coefficients ⁇ first again by masking out the bit fields shown in FIG. 20 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 is indicated in FIG. 18 with the multiplexer M and, in particular, has the consequence that the tables denoted by tanh and ⁇ tanh of the tangent hyperbolic -Function must only be saved once.
  • an acoustic signal x to be processed is completely in the time domain is processed.
  • a control variable ⁇ that is characteristic of the subjective loudness perception of normal hearing is calculated.
  • the input signal x is processed with a time-dependent filter 7, the parameters of which are continuously determined with the aid of the control variable ⁇ by interpolation of user-specific correction data calculated in advance 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 iteratively calculating the control variable ⁇ and a correction filter stage 7, which is thus controlled as a function of time.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Circuit For Audible Band Transducer (AREA)
EP97810460A 1996-07-19 1997-07-11 Traitement d'un signal acoustique basé sur le contrôle de l'intensité sonic Expired - Lifetime EP0820212B1 (fr)

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CH182396 1996-07-19
CH1823/96 1996-07-19

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

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

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DE59713033D1 (de) * 1996-07-19 2010-06-02 Bernafon Ag Lautheitsgesteuerte Verarbeitung akustischer Signale
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

Cited By (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
AU729074B2 (en) * 1996-07-19 2001-01-25 Bernafon Ag Loudness-controlled processing of acoustic signals
CN1640190B (zh) * 2001-08-08 2010-06-16 Gn瑞声达公司 使用数字频率扭曲的动态范围压缩
EP1404152A3 (fr) * 2002-09-30 2006-11-29 Siemens Audiologische Technik GmbH Dispositif et procédé d'adaptation d'une prothèse auditive
US7236603B2 (en) 2002-09-30 2007-06-26 Siemens Audiologische Technik Gmbh Device and method to adapt a hearing device
FR3052951A1 (fr) * 2016-06-20 2017-12-22 Arkamys Procede et systeme pour l'optimisation du rendu sonore de basses frequences d'un signal audio
WO2017220906A1 (fr) * 2016-06-20 2017-12-28 Arkamys Procede et systeme pour l'optimisation du rendu sonore de basses frequences d'un signal audio
CN110036653A (zh) * 2016-06-20 2019-07-19 阿嘉米斯 用于优化音频信号的低频声学再现的方法和系统
CN109998774A (zh) * 2017-12-22 2019-07-12 大北欧听力公司 具有多频带限制器的听力保护装置

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Publication number Publication date
EP0820212B1 (fr) 2010-04-21
AU729074B2 (en) 2001-01-25
US6370255B1 (en) 2002-04-09
DK0820212T3 (da) 2010-08-02
EP0820212A3 (fr) 2006-03-22
AU2865597A (en) 1998-01-29
DE59713033D1 (de) 2010-06-02

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