EP1195076A2 - Prothese auditive a format acoustique - Google Patents

Prothese auditive a format acoustique

Info

Publication number
EP1195076A2
EP1195076A2 EP00942779A EP00942779A EP1195076A2 EP 1195076 A2 EP1195076 A2 EP 1195076A2 EP 00942779 A EP00942779 A EP 00942779A EP 00942779 A EP00942779 A EP 00942779A EP 1195076 A2 EP1195076 A2 EP 1195076A2
Authority
EP
European Patent Office
Prior art keywords
signal
filter
hearing
output
hearing aid
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.)
Withdrawn
Application number
EP00942779A
Other languages
German (de)
English (en)
Inventor
Walter P. Sjursen
Geary A. Mccandless
Frederick J. Fritz
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.)
Sarnoff Corp
Original Assignee
Sarnoff Corp
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 Sarnoff Corp filed Critical Sarnoff Corp
Publication of EP1195076A2 publication Critical patent/EP1195076A2/fr
Withdrawn legal-status Critical Current

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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
    • H04R25/502—Customised settings for obtaining desired overall acoustical characteristics using analog 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
    • 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

  • One form of hearing loss is caused by reduced sensitivity at high frequencies.
  • This type of hearing impairment is sometimes corrected using frequency equalization.
  • the range of frequencies in which a listener has reduced sensitivity is amplified so that there is an even sensitivity of all frequencies across the audible range.
  • reduced sensitivity is more predominant at the high end of the audible frequency range.
  • frequencies at the high end of the audible spectrum are usually amplified to support equalization of sound across the audible spectrum for impaired listeners.
  • Such hearing aids are typically custom fitted to an individual based upon his or her audiological and physical needs. Accordingly, an acoustical format is created to compensate for the hearing loss of the individual.
  • a common method of providing a desired acoustical format is to create a custom ear mold or shell, which is made to fit the ear and/or ear canal of the individual. It can be a painstaking process to correct hearing using such methods because types of hearing impairments vary from one individual to another as do the physical characteristics of one ear canal versus another.
  • a basic hearing aid style such as behind-the-ear (BTE), in-the-ear (ITE), in-the-canal (ITC), or completely-in-the-canal (CIC) is selected for fitting.
  • BTE behind-the-ear
  • ITE in-the-ear
  • ITC in-the-canal
  • CIC completely-in-the-canal
  • This process might include adjusting mechanical features of the hearing aid such as switches or rotating trimmer controls.
  • the process might involve adjusting the characteristics of the device electronically using a programming device such as a handheld programming unit or computer interface. Whether proper fitting of an earpiece requires adjusting programmable circuits or the physical characteristics of the earpiece itself, the additional time to properly fit an individual with such devices results in an increase in cost without a substantial increase in benefit to the hearing impaired patient.
  • Analog signal processing for hearing aid applications initially consisted of frequency-independent linear amplification. Later, frequency compensation and compression circuits were included in the signal processing functions. In some hearing aid circuits, the frequency spectrum is split into two channels, a low- frequency channel and a high-frequency channel, with the gain and compression of each channel independently controlled.
  • Each channel includes filters.
  • Filters for analog signal processing are often implemented using op-amps.
  • the constraints of small size, low power and low operating voltage have driven the signal processing filters to rather simple filter designs.
  • the basic dynamic filter in a K-amp hearing aid circuit is implemented as a single op-amp, first order filter.
  • the Gennum DynamEQ hearing aid circuit in which the filter is implemented using a two op-amp first-order filter.
  • High order filters should provide the hearing aid user higher benefit, particularly in terms of a more natural sound since high order filters offer more spectral control. For example, high order filters can be tuned not to over-amplify frequencies where the user's residual hearing is still acceptable.
  • high order filters can be realized with a single op-amp, but for practical reasons, are often not.
  • a second-order filter section is often implemented using the well-known biquad configuration having three op-amps.
  • a typical, continuous-time, biquad, second-order filter section uses three op-amps, while a switched-capacitor version of the same biquad filter section can be implemented using only two op-amps. Therefore, analog signal processing with high-order filters often require many op-amps.
  • signal processing in a hearing aid application using sixth-order filters may use as many as eighteen or more op-amps.
  • hearing aids provide amplification to compensate for hearing loss. Under noise-free conditions, simple amplification and frequency compensated amplification systems provide acceptable performance. However, under noisy conditions and particularly conditions where the noise contains higher low-frequency components, the amplifiers of certain hearing aids are often driven into saturation (clipping). When clipping occurs, the high frequency components essential to speech intelligibility are sometimes so distorted or attenuated that high frequency components of the original signal disappear altogether.
  • some hearing aids include compression circuits. When loud low-frequency signals are present, these compression circuits reduce the gain of the hearing aid. This reduces the strength of the high-frequency signal components critical to speech intelligibility. Experience has shown, therefore, that these types of hearing aids are often ineffective in noisy environments such as restaurants, automobiles, trains and airplanes where low-frequency noise is predominant.
  • the present invention is generally directed towards certain aspects of hearing aid devices.
  • One aspect of the present invention involves segregating types of hearing impairments into classes or ranges. For each range, an approximate or average level of reduced sensitivity or hearing loss is determined. Accordingly, an acoustical format having a defined frequency response is then calculated to remedy hearing loss for the approximated types of hearing impairments for each range.
  • each hearing loss range covers a span of about 10-12 dB, reducing the number of ranges to a reasonably manageable number.
  • Hearing aids are preferably programmed at a factory with a fixed acoustical format and are prescribed to a patient for correcting a corresponding type of hearing impairment.
  • a set of predetermined parameters are used to define a matrix of hearing aids in which each of the hearing aids has one fixed acoustical format.
  • One parameter classifies a hearing aid according to a relative change of gain in a predetermined frequency range of an acoustical format.
  • a second parameter is used to further classify a hearing aid based on, for example, a maximum or peak gain of a particular acoustical format at a predetermined audible frequency.
  • the hearing aids are classified to produce a two dimensional matrix.
  • Each type of hearing aid in the matrix has a pre-programmed acoustical format or frequency response to remedy a hearing impediment depending on a severity of the corresponding type of hearing loss being corrected.
  • a transfer function provides basic characteristics of a 2-channel system.
  • An analog filter provides the flexibility to easily select the maximum slope of the transfer function between the low-frequency and high-frequency ranges. Slope selection flexibility facilitates, for example, hearing aid designs that can be tuned to offset the various hearing loss characteristics.
  • analog filter performance is improved and physical size of the high order filters is reduced.
  • second order filters are implemented using a combination of two op-amps and switched-capacitors.
  • higher order filters which include multipliers, produce the characteristics of the transfer functions operating in the hearing aids, for example.
  • a variable gain op-amp circuit is configured to act as a multiplier, which eliminates multiplier circuits to improve power savings further.
  • a minimum number of op-amps In an application such as hearing aids, it is useful to implement the signal processing design using a minimum number of op-amps.
  • the benefits of having a minimum number of op-amps include smaller size, lower noise, lower power, and lower cost when compared to implementations using a higher number of op-amps.
  • One aspect of the present invention is directed towards reducing the effects of background noise of an audio signal.
  • frequency components of an audio input signal are segregated and processed in separate channels.
  • a summer circuit is used to recombine the separated components into the output signal for a listener. Since separate channels are used to process corresponding frequency bands, clipping due to background noise in one channel does not completely deteriorate the integrity of frequency components of the other channels of the input signal.
  • the amplification of frequency ranges of a given channel is adjusted to compensate for the hearing loss of a hearing impaired patient.
  • Each channel includes a filter of which a corresponding output is fed into a non- linear amplifier.
  • the output of the amplifier is then fed into a second filter whose output is recombined at a summer circuit with the channel outputs of other frequency components of the original audio input signal.
  • the filter circuits for each channel are chosen so that the range of frequencies are contiguous across the audible spectrum. Based on this topology, it is possible to process an input signal to produce a related output signal at the summer circuit to compensate for the hearing loss of an individual.
  • Fig. 1 is a graph of hearing loss versus frequency which is representative of a range of hearing loss generally considered as mild to moderate.
  • Fig. 2 is a graph as in Fig. 1 showing the representative range of hearing loss segregated into a limited number of regions.
  • Fig. 3 is a graph as in Fig. 1 of the nominal hearing loss for each of the representative regions of hearing loss useful in understanding the principles of the present invention.
  • Fig. 4 is a graph plotting the gain in decibels versus frequency of representative target responses for a family of linear hearing aids which could compensate for the hearing loss plotted in Fig. 3 according to the present invention.
  • Fig. 5 is a graph as similar to Fig. 4 of the representative target responses for a family of non-linear hearing aids according to the present invention.
  • Fig. 6 is a block diagram illustrating functional components of the present invention.
  • Figs. 7 - 12 are magnitude Bode plots of transfer functions describing analog filters for use in hearing aid applications.
  • Fig. 13 is a block diagram of a generic analog filter for implementing a transfer function of the form depicted in Figs. 7-12.
  • Fig. 14 is a schematic diagram of a continuous-time biquad band-pass filter.
  • Fig. 15 is a schematic diagram of a switched-capacitor biquad band-pass filter.
  • Fig. 16 is a block diagram of circuitry to implement signal processing transfer function.
  • Fig. 17 is a block diagram of optimized circuitry to implement signal processing transfer function.
  • Fig. 18 is a schematic diagram of preferred implementation of this invention.
  • Fig. 19 is an example of a matrix of hearing aids classified according to the principles of the present invention.
  • Fig. 20 is a schematic diagram of a hearing aid utilizing separate channel circuits to process corresponding frequency components of an audio input signal.
  • Fig. 1 is a graph depicting a range of hearing loss considered as mild to moderate.
  • line A represents a reduced sensitivity of a hearing impaired individual having a mild case of hearing loss.
  • Line B represents a reduced sensitivity of a hearing impaired individual having a moderate case of hearing loss.
  • Shaded region 100 represents a continuum of hearing loss somewhere between the severity of hearing loss as represented by line A and line B.
  • line A and line B generally have a common shape to the extent that hearing loss is greater at high frequencies.
  • sensitivity of a patient having a hearing impairment as depicted by line B has a steeper hearing loss drop-off in the mid-range of audible frequencies as shown in Fig.l.
  • Fig. 2 is a graph illustrating ranges of hearing loss according to the principles of the present invention.
  • a first region 210 depicts a milder form of hearing loss and is defined as the region between line A and line C.
  • a second region 220 is the next more severe range of hearing loss and is defined as the region of hearing loss between line C and line D.
  • a third region 230 is the most severe range of hearing loss in our example and is defined as the region between line B and line D.
  • Each region e.g., the first region 210, second region 220, or third region 230, defines a class of hearing impaired individuals. Those within a class are considered to have very nearly the same type of hearing impairment.
  • Fig. 3 illustrates the same features as found in Fig. 2, however, several lines have been added depicting the average or approximated hearing loss for a specified region.
  • the approximate hearing loss for the first region 210 is defined by line X.
  • line Y is the approximated hearing loss for the second region 220
  • line Z is the approximated hearing loss for the third region 230.
  • Fig. 4 is a graph illustrating a desired hearing aid response for each of the hearing impediment ranges as defined by line X, line Y, and line Z of Fig. 3. As shown, the various frequency responses define a family of linear hearing aids. Acoustical format 410 provides appropriate gain at co ⁇ esponding frequencies of a received acoustical signal to restore hearing for individuals diagnosed with hearing impediments as defined by line Z.
  • acoustical format 410 to restore their hearing.
  • individuals diagnosed with hearing impediments in second region 220 are assigned an acoustical format 420 to correct their hearing.
  • individuals diagnosed with hearing impediments in the first region 210 are assigned an acoustical format 430 to correct their hearing.
  • the gain at a particular frequency as illustrated by the acoustical formats is about half that of the hearing loss. Accordingly, each acoustical format is about 5-6 dB apart from each other.
  • the hearing aids are accurate within 3 dB of the desired response. That is, the frequency response of an acoustical format due to manufacturing tolerances is off by no more than 3 dB from the ideal response as shown in Fig. 4 or Fig. 5.
  • Fig. 5 is a graph illustrating prefe ⁇ ed frequency responses for a family of non-linear hearing aids. For example, a desired frequency response of several acoustical formats is shown for each of the hearing impediments, as defined by line X, line Y, and line Z of Fig. 3.
  • An acoustical format 510 provides an appropriate gain for restoring normal hearing to individuals diagnosed with a hearing impediment as defined by line Z.
  • acoustical format 510 individuals diagnosed with hearing impediments that fall within the third region 230 are assigned acoustical format 510 to restore their hearing.
  • individuals diagnosed with hearing impediments within the first region 210 and the second region 220 are assigned acoustical format 530 and acoustical format 520, respectively, to co ⁇ ect corresponding types of hearing loss.
  • Fig. 6 is a block diagram illustrating the functional components of one embodiment of the present invention.
  • acoustical vibrations 605 in air are processed by a hearing aid 600 to provide restoration of hearing for a patient at ear canal 660.
  • acoustical vibrations 605 are detected at a microphone 610 that, in turn, produces a low-level electrical signal 615 co ⁇ esponding to the detected sound.
  • a pre-amplifier and compressor 620 amplify the low-level signal and compress it to fit within the dynamic range of audible hearing.
  • Amplified signal 625 at the output of the pre-amplifier and compressor 620 is then fed into an amplifier/filter circuit 630 for further processing.
  • the electrical characteristics of the filter/amplifier circuit 630 depend on an acoustical format type 627 selected.
  • One format type, for example, that can be selected for hearing aid 600 is acoustical format 530 as previously discussed.
  • the filter/amplifier 630 processes amplified signal 625 to produce an output signal 635, which is fed to an output driver 640 for driving a speaker 650. Sound output 655 from the speaker 650 is then directed to a patient's ear canal 660.
  • a single hearing aid 600 can be formatted to correct a particular type of hearing loss based on a selected acoustical format.
  • the shape of the frequency response is one aspect of an acoustical format and is identified by a code.
  • the code is a letter of the alphabet.
  • the code defines the steepness of the gain profile in a portion of the audible frequency band.
  • Peak gain information such as maximum gain at a certain frequency is another aspect of a particular acoustical format and is optionally identified by a number. For example, if a code indicates a particular shape of the frequency response curve, a number is optionally used to indicate peak gain information of an acoustical format type or possibly the range of gains of a particular acoustical format. In one embodiment, the number refers to the maximum or peak gain in decibels (dB).
  • dB decibels
  • hearing aids are classified based on steepness of a frequency response and degree of hearing loss at a particular frequency.
  • the classes of different types of hearing loss are equally spaced. That is, the hearing loss ranges are preferably separated by equal spacings such that the hearing loss of different classes at a predetermined frequency is 28, 40, 52, 64, and 84 dB.
  • curve shape, gain range or peak gain are sufficient to accurately describe many different types of acoustical formats. This renders it possible for a hearing aid provider to maintain a limited number of acoustical formats while providing hearing aids for many different types of hearing loss.
  • each hearing aid is programmed at a factory with a predetermined acoustical format and is not re-programmable.
  • a matrix of different types of hearing aids each having a different acoustical format are then maintained at, for example, a local pharmacy.
  • hearing aids in the matrix are low- cost and disposable.
  • an appropriate hearing aid is selected from the matrix of hearing aids and prescribed by an audiologist, the patient need only pick-up the prescription at a local pharmacy supplying such devices. Based on this method, there is no need to make adjustments to the hearing aid at the audiologist's office. Rather, a patient's type of hearing loss is identified by the audiologist and the corresponding type of hearing aid is prescribed to remedy the hearing impairment.
  • One aspect of the invention describes an analog filter suitable for hearing aid applications.
  • the analog filter is described in the s-domain with the following normalized equations:
  • V(s) ((s-l) / (s+l)) n (2)
  • X(s) N(s) - U(s) (3)
  • s is the complex frequency j ⁇
  • ⁇ controls the resonance of the second-order filter section U(s)
  • n selects the number of sections and, hence, the maximum slope of the filter.
  • Parameter n is an integer normally in the range of 1 to 4.
  • U(s) defines a high-pass filter
  • N(s) defines an all-pass filter
  • X(s) defines a low-pass filter.
  • U(s) and X(s) are combined in various ratios to produce the desired transfer function of the hearing aid as follows:
  • (1+ ⁇ ) is the high-frequency gain and (1+ ⁇ - ⁇ ) is the low frequency gain.
  • the parameter ⁇ controls the filter and, in particular, the high-frequency gain while parameter ⁇ controls the amount of low- frequency gain relative to the high- frequency gain.
  • Equations 1-4 describe a family of transfer functions suitable for hearing aid applications.
  • f c , ⁇ , n, a and ⁇ By varying f c , ⁇ , n, a and ⁇ , a wide range of transfer functions suitable for hearing aid applications can be achieved.
  • Figures 7-12 show representative frequency responses for transfer function T(s) for some different values of f c , ⁇ , n, a and ⁇ .
  • the independent variables, ⁇ , ⁇ , n defining the transfer function, T( ⁇ , ⁇ , n), produce the family of frequency responses.
  • n 1
  • a block diagram of the analog filter described by equations (1) through (4) above is shown in Fig.13.
  • the signal path is shown as solid lines and the control signals are shown as dashed lines.
  • An automatic gain control (AGC) circuit generates the control signal alpha ( ⁇ ) based on characteristics of the signal U(s).
  • This analog filter includes three second-order high-pass filter sections and three first-order all-pass filter sections. It should be understood that the analog filter comprises analog components (e.g., resistors and capacitors) that relate to parameters ⁇ , ⁇ , ⁇ and n.
  • the analog filter described above and defined by equations (1) through (4) is replaced with an analog filter that replaces the high-pass filter sections with band-pass filter sections and eliminates the all-pass filter sections.
  • the alternate analog filter is suitable for hearing aid applications and its transfer function is described in the s-domain with the following normalized equations:
  • U(s) defines a band-reject filter (i.e., a notch filter)
  • X(s) defines a band-pass filter
  • T(s) defines the overall filter transfer function.
  • the parameter ⁇ controls the high-frequency gain
  • a controls the low-frequency gain relative to the high-frequency gain
  • ⁇ controls the sharpness of the band-pass filter
  • ⁇ controls the maximum high frequency gain in conjunction with ⁇ .
  • the parameter n defines the number of cascaded band-pass filters, where n is normally in the range from one to three.
  • the ⁇ parameter may take on a range from 1 to 2 and more preferably between 1.4 and 1.6.
  • the ⁇ parameter has a value of 1.538 (i.e., 1/0.65). While equations (5) through (7) have been normalized to a characteristic frequency of 1 radian sec, one skilled in the art will realize that a much higher characteristic is needed for a hearing aid application.
  • the characteristic frequency will be scaled to between 3000 Hz (18850 radian/sec) and 7000 Hz (43982 radian/sec). In the present embodiment of the invention, the characteristic frequency is scaled to 5000 Hz (31461 radian/sec).
  • the following description presents an electronic circuit providing a configurable high-order filter primarily for hearing aid applications, such as for generating transfer functions (5) - (7) described above.
  • the electronic circuit embodies a filter that generally provides high-frequency amplification relative to low frequencies.
  • the prefe ⁇ ed embodiment of the invention uses less circuitry and, in particular, fewer op-amps (operational amplifiers) than non-prefe ⁇ ed embodiments.
  • Fig. 14 shows a schematic diagram of the well known continuous-time, second-order biquad band-pass filter. As shown in the figure, two band-pass outputs exist. One is a non-inverting band-pass output while the other is an inverting band-pass output.
  • Fig. 15 shows a schematic diagram of the same biquad band-pass filter implemented using switched-capacitor resistors.
  • the resistance may be either positive or negative (i.e., inverting switched-capacitor resistor). Since negative resistors can be implemented, the switched-capacitor biquad band-pass filter can use one fewer op-amp than the continuous-time filter of Fig. 14. Also, for the switched-capacitor biquad filter, only one band-pass filter output is available. To make either an inverting or non-inverting band-pass filter, the input resistor can be made either non-inverting or inverting, respectively. This is shown in Fig. 15 as resistors Rl-A (negative resistance for non- inverting band-pass output) and Rl-B (positive resistance for inverting band-pass output).
  • Fig. 16 shows a block diagram of the desired signal-processing algorithm described by equations (5) - (7) listed and described above.
  • the circuit includes a band-reject filter and three band-pass filters.
  • the output of the band-pass filters are X 1 , X 2 , - dX 3 , respectively.
  • a selector i.e., multiplexer
  • the output of the selector is designated X".
  • the output of the selector goes to an automatic gain control (AGC) control circuit, which develops a control signal designated alpha.
  • the signal alpha is multiplied by a constant factor gamma to generate the control signal designated alpha*gamma.
  • AGC automatic gain control
  • the output of the band-reject filter is multiplied by a constant factor beta to generate a signal designated beta*U.
  • the signal X" is subtracted from beta* U and multiplied by al ⁇ ha*gamma to create a signal designated alpha*gamma*(beta*U-X") + U.
  • the band-reject signal -7 is added to alpha*gamma*(beta*U-X) to create the output signal alpha*gamma*(beta*U-X n ) + U.
  • a straightforward implementation of the system shown in Fig. 16 uses at least 13 op-amps, excluding the AGC control circuit. This is assuming switched- capacitor filters using 2 op-amps for each band-pass filter and 3 op-amps for the band-reject filter. For low-power applications such as hearing aids, it is desirable to minimize the number of op-amps. By using fewer op-amps, three improvements are achieved: (1) less power is needed, (2) less silicon area is needed for a custom integrated circuit, and (3) less silicon area translates into lower cost.
  • the invention describes a prefe ⁇ ed embodiment of the signal-processing algorithm, shown in Figs. 17 and 18, in which only 9 op-amps are needed.
  • Fig. 17 shows a block diagram of the prefe ⁇ ed embodiment of the signal- processing design.
  • the circuit includes three band-pass filters.
  • One band-pass filter is an inverting band-pass filter, while the other two band-pass filters are non- inverting.
  • the outputs of the three band-pass filters are designated - ' , -X 2 , and -X respectively.
  • a selector selects one of the band-pass filter outputs based on a control signal, designated n.
  • the output of the selector is designated -X".
  • the output of the selector goes to an AGC control circuit, which develops a control signal designated alpha.
  • An inverting summing amplifier sums the output of the selector -X", the output of the first (inverting) band-pass filter -X 1 weighted by a constant factor beta, and the input signal also weighted by a constant factor beta, to form an output signal that is weighted by another constant factor gamma and designated by - gamma* (beta*U-X").
  • the output of this first inverting summing amplifier goes through an inverting amplifier, with a gain factor controlled by alpha to generate an output signal designated alpha* gamma* (beta*U-X").
  • a second inverting summing amplifier sums the output of said inverting amplifier with the output of the first (inverting) band-pass filter and the input signal to generate an output signal designated -(alpha*gamma*(beta*U-X n ) + U).
  • the output signal of the block diagram of Fig. 17 is identical to the output of the block diagram of Fig. 16.
  • the final inverting summing amplifier in Fig. 17 is replaced with a non-inverting summing amplifier, and the output signal is designated (alpha*gamma*(beta*U-X") + U) (i.e., the leading negative sign is deleted).
  • Fig. 18 shows a schematic diagram of the prefe ⁇ ed embodiment of the present invention.
  • the circuit implements the signal-processing design shown in Fig. 17.
  • the first (inverting) band-pass filter comprises op-amps AR101 and AR102, resistors R101-R104 and capacitors C101 and C102.
  • the second (non- inverting) band-pass filter comprises op-amps AR201 and AR202, resistors R201- R204 and capacitors C201 and C202.
  • the third (non-inverting) band-pass filter comprises op-amps AR301 and AR302, resistors R301-R304 and capacitors C301 and C302.
  • the selector comprises switches SI -S3.
  • the first inverting summing amplifier comprises op-amp AR1 and resistors R1-R5, where resistor R3 sets the constant factor beta, and resistor R5 sets the constant factor gamma.
  • the inverting amplifier comprises op-amp AR2 and resistors R6 and R6, where by varying the resistance of either resistor R6, R7, or both R6 and R7 varies the amplification factor alpha.
  • the second inverting summing amplifier comprises op-amp AR3 and resistors R8-R11.
  • the circuit of Fig. 15 comprises a total of nine op-amps. Although component values (resistors and capacitors) are not shown, one skilled in the art can easily determine a set of component values to achieve the desired signal- processing algorithm.
  • op-amp AR3 and resistor Rl 1 of Fig. 18 may be eliminated.
  • the final inverting summing amplifier is replaced with the resistive summing network comprising resistors R8-R10.
  • This embodiment of the invention uses only eight op-amps.
  • the output signal, taken at the junction of resistors R8-R10, is given by (113)* ((alpha* gamma* (beta* U-
  • a matrix of hearing aids is preferably defined by attributes of the hearing device.
  • defining a class of hearing aids includes separating types of hearing aids based on steepness of response gain in a range of frequencies and peak response gain at a predetermined frequency.
  • Fig. 19 is an example of a hearing aid matrix where hearing aids are classified according to their frequency response characteristics as described above.
  • the 3 X 3 matrix classifies 9 different types of hearing aids.
  • Each hearing aid is preferably pre-programmed at a factory with a unique acoustical format for remedying a certain type of hearing loss.
  • a hearing aid is programmed with multiple types of acoustical formats, while only one of the acoustical formats is selected at a time.
  • Hearing aids in a column such as F-20, F-26 and F-32, define a class of hearing aids having a similar frequency response characteristic but different peak gain values.
  • "F” in the hearing aid identifier co ⁇ esponds to "flat,” which describes the frequency response of a particular class of devices. For example, see acoustical format 510 as shown in Fig. 5. The steepness of the gain slope in mid-range frequencies 1000-1200 Hertz is relatively flat and, therefore, the acoustical format 510 would be classified accordingly as a class "F" type of hearing aid.
  • the letter “S” in our exemplary 3 X 3 matrix stands for “steep,” while the letter “P” stands for “precipitous” (very steep).
  • classes of acoustical formats like acoustical format 520 (in Fig. 5) having a steep gain slope in mid-range frequencies 1000-1200 Hertz are assigned the letter "S.”
  • classes of acoustical formats like acoustical format 510 (in Fig. 5) having a very steep gain slope in the mid-range frequencies 1000-1200 are assigned the letter "P.”
  • acoustical formats are generated as shown above using the desired filters to create a target response as shown in Figs. 7-12.
  • a second parameter is used to further distinguish hearing aids having the same assigned letter.
  • hearing aids having a flat response i.e., F-20, F-26 and F-32, as shown in Fig. 19 include respective numerals, i.e., 20, 26, and 32, co ⁇ esponding to the peak gain (in decibels) of the frequency response of a particular type of hearing device.
  • acoustical formats 510, 520 and 530 as shown in Fig. 5 could be classified as P-32, S-26 and F- 20 respectively.
  • peak gain for this family of target responses corresponds with numerical value of the hearing aid, i.e., 32, 26 and 20.
  • acoustical format 510 It has a precipitous (very steep) gain slope in the middle of audible frequency range and a peak gain at 8000 hertz of 32 decibels.
  • this hearing aid would be classified in the matrix as P-32.
  • Hearing aids programmed with acoustical format 520 having a steep gain slope and a peak gain of 26 decibels would be classified in the matrix as S-26.
  • hearing aids programmed with acoustical format 530 having a flat gain slope and peak gain of 20 would be classified in the matrix as F-20.
  • Fig. 20 is a schematic diagram of a hearing aid device utilizing separate channels to process frequency components of an audio input signal.
  • Microphone 255 detects acoustical vibrations and produces an audio input signal 257 that is fed to each of multiple Channels 1 through N.
  • N is an integer greater than 1.
  • Each channel includes a co ⁇ esponding bandpass filter 250-1 (Channel 1), 250-2 (Channel 2)...250-N (Channel N) to separate audio input signal 257 into bands of frequency components.
  • the bandpass filter 250-1 passes a band of lower frequencies such as 100-500 Hz (Hertz) for signal processing in channel 1
  • bandpass filter 250-2 passes frequencies such as 500-1000 Hz
  • bandpass 250-N passes a band of higher frequency components such as 10-12 KHz (Kilohertz) for signal processing in channel N.
  • frequency components of the audio input signal 257 are separated so that they can be processed individually. Accordingly, distortion caused by clipping in one channel will not effect the integrity of frequency components processed by the other channels.
  • each channel's bandpass filters 250 is fed into a co ⁇ esponding non-linear amplifier 260 for a particular channel 1 through N.
  • the non-linearity of the amplifier is optionally implemented using either hard-clipping or soft-clipping. When nonlinear amplifiers are utilized, soft clipping is prefe ⁇ ed because distortions produced by the amplifier will generally be less damaging than when hard-clipping techniques are used.
  • nonlinear amplifiers 260 are also programmed to provide the appropriate gain to compensate for the hearing loss of a hearing impaired patient. For example, if a patient has hearing loss in a particular frequency range, the gain of the amplifier 260 is adjusted for altering the components of the original audio signal so that an impaired patient hears as if he had more normal hearing.
  • each channel has a second bandpass filter 270 that matches the characteristics of the first bandpass filter 250.
  • the inclusion of the second bandpass filter 270 is beneficial because it helps to reduce unwanted frequency components such as noise outside the bandpass of the channel, thus, producing a purer output.
  • Output signals of the second bandpass filter 270-1 through 270-N are fed into summer circuit 285 to drive a sound producing device such as a speaker.
  • linear amplifiers are optionally used to provide signal gain for a particular channel.
  • components of fig. 20 are optionally embodied using analog circuitry rather than digital signal processors and related circuitry.
  • the previously described hearing aid has many advantages. For example, amplifier distortions caused by loud low frequency background noise will not effect high-frequency components of the audio input signal because they are separated by individually processed channels. Clipping in one channel, therefore, will not effect the performance of the other channel. Typically, low frequency background noise is responsible for clipping in amplifiers. Since the high-frequency channel is not clipped along with the low- frequency channel as found in previous applications, speech intelligibility is preserved. That is, high-frequency components are preserved for the listener even though there is distortion in one of the channels.
  • Another advantage is the minimal circuitry required to create a low-cost hearing aid.
  • the packaging of the circuit is minimal and therefore less invasive to a patient wearing the hearing aid.
  • Certain aspects of the present invention have been discussed in terms of a hearing aid application. However, such principles are optionally used in communication systems in which speech needs to be transmitted in the presence of noise and, in particular, low-frequency noise. For example, applications such as cellular telephones can potentially benefit from the principles of the present invention by reducing the effects of low-frequency "road” or background noise when using the telephone in a vehicle such as an automobile, bus or train.

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)
  • Filters That Use Time-Delay Elements (AREA)
  • Electric Propulsion And Braking For Vehicles (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

Les types de perte auditive sont répartis en un nombre de classes prédéterminé. Pour chaque classe, on détermine une perte auditive approximative, et un format acoustique fixe correspondant. On affecte, par exemple, une réponse en fréquence afin de corriger un type de déficience auditive correspondant. Les prothèses auditives sont ensuite classées dans une matrice en fonction des caractéristiques de réponse en fréquence correspondante. On obtient une réponse en fréquence de prothèse auditive appropriée à l'aide de filtres analogiques, utilisant éventuellement des formes de condensateur commuté.
EP00942779A 1999-06-15 2000-06-13 Prothese auditive a format acoustique Withdrawn EP1195076A2 (fr)

Applications Claiming Priority (7)

Application Number Priority Date Filing Date Title
US13920499P 1999-06-15 1999-06-15
US139204P 1999-06-15
US15797399P 1999-10-06 1999-10-06
US157973P 1999-10-06
US52404300A 2000-03-13 2000-03-13
US524043 2000-03-13
PCT/US2000/016193 WO2000078096A2 (fr) 1999-06-15 2000-06-13 Prothese auditive a format acoustique

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EP1195076A2 true EP1195076A2 (fr) 2002-04-10

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EP00942779A Withdrawn EP1195076A2 (fr) 1999-06-15 2000-06-13 Prothese auditive a format acoustique

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EP (1) EP1195076A2 (fr)
JP (1) JP2003501986A (fr)
AU (1) AU5735200A (fr)
TW (1) TW465248B (fr)
WO (1) WO2000078096A2 (fr)

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WO2008141672A1 (fr) * 2007-05-18 2008-11-27 Phonak Ag Procédure de réglage pour dispositifs auditifs et dispositifs auditifs correspondants
US8412495B2 (en) * 2007-08-29 2013-04-02 Phonak Ag Fitting procedure for hearing devices and corresponding hearing device
CN101868983A (zh) * 2007-11-22 2010-10-20 索内提克有限公司 用于提供助听器的方法和系统
RU2462831C2 (ru) * 2007-11-22 2012-09-27 Сонетик Аг Способ и система обеспечения слуховым аппаратом
US8571245B2 (en) 2009-06-16 2013-10-29 Panasonic Corporation Hearing assistance suitability determining device, hearing assistance adjustment system, and hearing assistance suitability determining method
EP2302952B1 (fr) 2009-08-28 2012-08-08 Siemens Medical Instruments Pte. Ltd. Auto-ajustement d'un appareil d'aide auditive
EP2426953A4 (fr) * 2010-04-19 2012-04-11 Panasonic Corp Dispositif d'installation d'aide auditive
DK2567552T3 (en) 2010-05-06 2018-09-24 Sonova Ag METHOD OF OPERATING A HEARING AND HEARING
JP5991923B2 (ja) * 2011-07-08 2016-09-14 パナソニック株式会社 補聴適合度判定装置、および、補聴適合度判定方法
CN102611977A (zh) * 2012-02-15 2012-07-25 嘉兴益尔电子科技有限公司 一种通用型助听功能初始放大曲线及滤波器参数配置方法
US9729982B2 (en) 2012-06-19 2017-08-08 Panasonic Intellectual Property Management Co., Ltd. Hearing aid fitting device, hearing aid, and hearing aid fitting method
TWI623234B (zh) * 2016-09-26 2018-05-01 宏碁股份有限公司 助聽器及其自動分頻濾波增益控制方法
US10779082B2 (en) * 2018-05-30 2020-09-15 Magic Leap, Inc. Index scheming for filter parameters

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US4484345A (en) * 1983-02-28 1984-11-20 Stearns William P Prosthetic device for optimizing speech understanding through adjustable frequency spectrum responses
US5881159A (en) * 1996-03-14 1999-03-09 Sarnoff Corporation Disposable hearing aid
US5915031A (en) * 1996-04-30 1999-06-22 Siemens Hearing Instruments, Inc. Modularized hearing aid circuit structure

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Also Published As

Publication number Publication date
AU5735200A (en) 2001-01-02
WO2000078096A2 (fr) 2000-12-21
JP2003501986A (ja) 2003-01-14
WO2000078096A3 (fr) 2001-12-06
TW465248B (en) 2001-11-21

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