EP0820212B1 - Lautheitsgesteuerte Verarbeitung akustischer Signale - Google Patents

Lautheitsgesteuerte Verarbeitung akustischer Signale Download PDF

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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
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Prior art keywords
filter
signal
value
interpolation
values
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EP97810460A
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German (de)
English (en)
French (fr)
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EP0820212A2 (de
EP0820212A3 (de
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 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.

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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 Lautheitsgesteuerte Verarbeitung akustischer Signale Expired - Lifetime EP0820212B1 (de)

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

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

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EP (1) EP0820212B1 (da)
AU (1) AU729074B2 (da)
DE (1) DE59713033D1 (da)
DK (1) DK0820212T3 (da)

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US6370255B1 (en) 2002-04-09
EP0820212A2 (de) 1998-01-21
DK0820212T3 (da) 2010-08-02
EP0820212A3 (de) 2006-03-22
AU2865597A (en) 1998-01-29
DE59713033D1 (de) 2010-06-02

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