Classifications

• • 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
  • H04R2225/41—Detection or adaptation of hearing aid parameters or programs to listening situation, e.g. pub, forest
• • 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
  • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
  • H04R2430/01—Aspects of volume control, not necessarily automatic, in sound systems
• • 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
  • H04R25/40—Arrangements for obtaining a desired directivity characteristic
  • H04R25/407—Circuits for combining signals of a plurality of transducers

Definitions

• the invention relates to a method for operating a hearing aid and to a hearing aid itself.
• the microphone comprises a microphone and a receiver.
• a hearing aid People with hearing loss typically use a hearing aid. This usually involves an electromechanical transducer that captures ambient sound. The resulting electrical (audio) signals are amplified by an amplifier circuit and then output via another electromechanical transducer in the form of a receiver, thus delivering the sound into the person's ear canal. The captured audio signals are usually also processed, typically by a signal processor within the amplifier circuit. The amplification is adjusted to the specific hearing loss of the hearing aid user, who is also referred to as the user or wearer.
• unprocessed ambient sound enters the person's ear canal in addition to the output sound.
• this unprocessed ambient sound enters, for example, between the hearing aid and the edge of the ear canal, or between the receiver inserted into the ear canal and the edge of the ear canal, allowing the person to perceive it as well.
• comb filter One such artifact is the so-called comb filter.
• minima in the amplitude appear at specific frequency intervals in the frequency spectrum of the superimposed, unprocessed ambient sound and the output sound, with these minima being relatively sharply defined. The person perceives this superposition as if they were in a tunnel.
• the audibility of the comb filter artifact depends strongly on the type and level of the input signal. Due to the non-linear amplification of hearing aids and user-specific adjustments, the comb filter artifact varies between individuals. Therefore, implementing solutions for this is difficult.
• Another method involves attenuating/not amplifying the relevant frequencies during operation only when the current situation suggests a high probability of comb filter artifacts occurring. This relies on estimating the superposition, which is comparatively computationally intensive and prone to errors. If the estimation is incorrect, other artifacts can occur, and intelligibility for the user may also be reduced.
• the invention is based on the objective of providing a particularly suitable method for operating a hearing aid as well as a particularly suitable hearing aid. to specify, in particular increasing user comfort and appropriately reducing hardware resources.
• the hearing aid is a headphone or includes a headphone, and the hearing aid is, for example, a headset.
• the hearing aid is most commonly referred to as a hearing aid device.
• the hearing aid device serves to support a person suffering from a reduction in hearing ability.
• the hearing aid device is a medical device by means of which, for example, partial hearing loss is compensated.
• the hearing aid device is, for example, a receiver-in-the-canal (RIC) hearing aid, an in-the-ear hearing aid such as an in-the-ear (ITC) or complete-in-the-canal (CIC) hearing aid, hearing glasses, or a pocket hearing aid.
• the hearing aid is a behind-the-ear (BTE) hearing aid, which is worn behind the ear.
• BTE behind-the-ear
• the hearing aid is designed and configured to be worn on the human body.
• the hearing aid preferably includes a retention device that allows it to be attached to the human body.
• the hearing aid is a hearing assistance device, it is designed and configured to be placed, for example, behind the ear or within an ear canal.
• the hearing aid is wireless and designed and configured to be inserted, at least partially, into an ear canal.
• the hearing aid includes a microphone that serves to capture sound. Specifically, when in operation, the microphone captures ambient sound, i.e., sound waves, or at least a portion thereof.
• the microphone is therefore appropriately The microphone is at least partially located within a housing of the hearing aid and is thus at least partially protected.
• the microphone is suitably an electromechanical transducer.
• the microphone may, for example, have only a single microphone unit or several microphone units that interact with each other.
• Each of the microphone units expediently has a diaphragm that is set into vibration by sound waves, the vibrations being converted into an electrical signal by means of a suitable recording device, such as a magnet moved within a coil.
• the microphone units are capacitively designed, and the fact that an applied electrical voltage changes when the distance between the diaphragm and a static surface of the microphone unit changes is utilized.
• the electrical voltage is applied, in particular, between the diaphragm and the static surface.
• the microphone units are preferably omnidirectional. In this or another way, it is at least possible to generate or at least provide an electrical signal using the microphone, which is based on the sound incident on the microphone, namely, in particular, ambient sound.
• the electrical signal constitutes an input signal.
• the hearing aid has a receiver for outputting a signal.
• the output signal is, in particular, an electrical signal, and may be, for example, digital or, more appropriately, analog.
• the receiver is preferably an electromechanical transducer, such as a loudspeaker.
• the receiver is at least partially located within the ear canal of a user of the hearing aid, i.e., a person also referred to as the wearer, user, or hearing aid wearer, or at least acoustically connected to it.
• the hearing aid primarily serves to output the signal via the receiver, thereby generating a corresponding sound.
• the main function of the hearing aid is preferably to output the signal, thus generating the sound.
• the hearing aid suitably includes a signal processing unit by means of which the microphone and the receiver are connected.
• the hearing aid has a signal processor that, for example, forms the signal processing unit or is at least a component thereof.
• the signal processor is, for example, a digital signal processor (DSP) or implemented using analog components.
• DSP digital signal processor
• the signal processor, or at least the signal processing unit is used in particular to adapt the input signal generated by the microphone, preferably to create the output signal.
• the signal processing unit is suitable for this purpose, and in particular, is designed and configured accordingly.
• an analog-to-digital converter (ADC) is arranged between the microphone and the signal processing unit, for example, the signal processor, provided the signal processor is designed as a digital signal processor.
• the hearing aid additionally includes an amplifier, or the amplifier is at least partially formed by the signal processing unit. For example, the amplifier is connected upstream or downstream of the signal processor.
• the method involves generating the input signal from the ambient sound.
• the ambient sound is captured, and the input signal is created from it.
• the input signal is preferably an electrical signal, and its generation is expediently carried out using the microphone(s).
• the input signal corresponds, for example, to the unprocessed ambient sound or may already be processed.
• the input signal expediently has a specific directional characteristic, so that a particular part of the environment is amplified, specifically sound from a certain solid angle.
• An output signal is generated from the input signal.
• the input signal can be directly mapped to the output signal.
• the input signal is at least partially processed, and the processed input signal corresponds, in particular, to the output signal.
• the processing is carried out, for example, in one step or in several steps. In particular, frequency-specific amplification is performed. and/or certain frequencies are mapped to other frequencies, resulting in compression. Alternatively or in combination with this, at least a partial delay is applied.
• the signal processing unit is used to provide the output signal, and preferably also to process it.
• frequency bands whereby all frequencies within the same frequency band are processed identically.
• the frequency bands are structured in such a way that the resulting discretization is imperceptible to the user.
• the output sound is then emitted via the receiver based on the output signal.
• the (electrical) output signal is converted into sound waves, for which the receiver is designed.
• the output sound is based on the output signal, and the output sound is emitted via the receiver when it receives the output signal.
• the output signal namely the output sound, is perceptible to the user.
• the output sound is directed into the user's ear canal.
• the method also determines the unprocessed ambient sound present in the area of the receiver.
• this area is advantageously located in the ear canal and preferably includes the eardrum.
• the unprocessed ambient sound is measured directly, for instance, using an additional microphone positioned in the area of the receiver.
• the unprocessed ambient sound is estimated, for which a transfer function is used in particular.
• the transfer function is specifically used to... An input signal is fed in.
• the transfer function is determined theoretically and/or arises solely from the design of the hearing aid. In this case, the transfer function is the same for all users. Alternatively, it may be specifically created for each user, for example, when the hearing aid is fitted to the individual user.
• each frequency band may contain only a single frequency, or preferably, each frequency band may be assigned adjacent frequencies.
• the same frequency bands are used that were also used to generate the output signal.
• the respective amplitude is taken into account in particular, and consequently, the ratio of the unprocessed ambient sound that penetrates the ear canal past the hearing aid to the possible processing by the signal processing unit, i.e., the output sound, is determined.
• the output sound is measured directly, for example.
• this sound is also determined theoretically, for which an additional transfer function is expediently used.
• This additional transfer function differs from any transfer function used to determine the unprocessed ambient sound.
• the additional transfer function depends on any processing of the input signal.
• the additional transfer function takes into account any frequency-selective amplification and/or a time offset, i.e., in particular, different propagation delays.
• the ratio is between an upper limit and a lower limit, the amplitude of the output signal is changed for this frequency band. This then changes the resulting ratio.
• a change in amplitude occurs when the ratio is at least approximately equal to 1, and when the amplitudes of the unprocessed ambient sound and the output sound are essentially the same.
• the potential artifacts resulting from superposition are comparatively pronounced, at least compared to a reduction at either amplitude. Due to the change in the output sound's amplitude, the strength of any resulting artifact is reduced, so that if present, the artifact is either imperceptible or only faintly noticeable to a person. This increases user comfort.
• the input signal can be processed in a way that is always adapted to the user's potential hearing loss, further increasing comfort. Furthermore, the process does not require any comparatively complex calculations, thus reducing the hardware resources needed.
• the processed ambient sound and the output sound are each measured to determine the ratio.
• this is performed for only a single frequency band.
• this is performed for multiple frequency bands.
• different frequency bands may have different upper/lower limits assigned to them. Preferably, however, these limits are the same for all frequency bands, thereby reducing complexity.
• the change in amplitude is, for example, always the same, or preferably dependent on the ratio, i.e., the value of the ratio. Alternatively, or in combination with this, the change in amplitude depends on other conditions, so that the current situation and/or different artifacts are taken into account. This further increases user comfort.
• the amplitude is set to a predefined value. For instance, the amplitude is increased. However, decreasing it is particularly preferred.
• this frequency range of the output signal is only perceived as reduced by the user via the corresponding output sound. Since the unprocessed ambient sound has an amplitude of the same order of magnitude in this frequency band, the corresponding frequency band remains perceptible. Therefore, the user is essentially unaware that the amplitude of the output signal has been reduced, and intelligibility remains high. This increases comfort.
• an upper limit of 20 dB or 10 dB is used.
• a particularly preferred upper limit is 6 dB.
• a lower limit of -20 dB or -10 dB is used, for example.
• a lower limit of -6 dB is used.
• an upper limit of 6 dB and a lower limit of -6 dB are used.
• the method only determines the amplitude for frequency bands with frequencies between 50 Hz and 5 kHz, between 100 Hz and 3 kHz, or between 200 Hz and 2.5 kHz, and modifies the amplitude if necessary.
• only the amplitude is modified for frequency bands with frequencies between 250 Hz and 2 kHz.
• the ratio is only determined for such frequency bands, and thus the respective transfer function is only used for the unprocessed ambient sound and/or the output sound. This reduces the effort.
• a comb filter artifact occurs predominantly at frequencies between 250 Hz and 2 kHz, or is at least bothersome to the user there. Therefore, the adjustment is only performed in the frequency range relevant to the user, while the unaltered output signal is used elsewhere. This reduces the effort without compromising user comfort.
• desired effects that were incorporated through the corresponding generation of the output signal are not unintentionally undone.
• the gain factor assigned to the frequency band used to generate the output signal is changed to alter its amplitude.
• the input signal is divided into different frequency bands to generate the output signal, with each frequency band being amplified by its own assigned gain factor.
• the assigned gain factor By changing the assigned gain factor, the resulting ratio is altered in the very step of generating the output signal, thus reducing the number of required steps and minimizing complexity.
• the amplification factor is increased. However, it is more practical to decrease it so that this portion of the output signal is in the It is essentially imperceptible. This avoids excessive noise levels, which could otherwise reduce comfort.
• the gain factor is only changed in the frequency band where the ratio between the upper and lower limits lies.
• the adjacent frequencies are also adjusted, preferably using a corresponding target function for this adjustment.
• This function is particularly notch-shaped. This ensures a smooth transition during the change, thus increasing ease of use.
• to change the gain factor it is multiplied by a predetermined number.
• a specific constant is added or subtracted.
• this constant is between 1 dB and 18 dB, suitably between 5 dB and 15 dB, and, for example, essentially equal to 10 dB.
• the ratio is also adjusted in such a way that no further adjustment is necessary, since the ratio lies outside the upper and lower limits.
• the gain factor is changed relatively abruptly when the ratio between the upper and lower limits is...
• This is specified, in particular, by means of a lookup table or a function.
• the target value is between 5 dB and 15 dB, and preferably equal to 10 dB. Due to the gradual adjustment, an abrupt change, which would be unpleasantly perceptible to the user, is avoided. Consequently, comfort is increased.
• the amplitude is always changed when the ratio between the upper limit and the lower limit is zero. Preferably, however, this only occurs when a criterion is met, i.e., an additional condition. This ensures that, for example, in situations where desired.
• the key is that the ratio between the upper and lower limits is constant, ensuring no change in amplitude occurs, which would otherwise negate a desired effect. This makes it possible to guarantee that adjustments are only made when unwanted artifacts are present or at least potentially present, thus increasing user comfort.
• the criterion may, for example, comprise one or more conditions, and if at least one is met, the criterion is fulfilled. Alternatively, it may be necessary for certain conditions or a specific number of conditions to be met for the criterion to be fulfilled. Alternatively, it may be necessary for all conditions to be met.
• the amplitude is changed solely based on whether the criterion is met.
• the criterion is also particularly useful for determining the extent of the amplitude change. This allows for the targeted removal of unwanted artifacts.
• the criterion is expediently met only if the amplitude is more than a third threshold smaller than a reference amplitude, i.e., if a notch is present in the frequency spectrum.
• the reference amplitude is, in particular, the average of all amplitudes in the frequency range or at least in the adjacent frequency bands.
• the criterion is met only if the amplitude is more than a fourth threshold larger than the respective reference amplitude, i.e., if a peak is present.
• the third and fourth thresholds are preferably of the same magnitude and, in particular, larger than 10 dB or at least larger than 5 dB.
• a change in amplitude essentially only occurs if a kind of comb-like structure is present, i.e., a comparatively sharp minimum or maximum.
• a change in amplitude essentially only occurs if a kind of comb-like structure is present, i.e., a comparatively sharp minimum or maximum.
• the amplitude is smaller/larger than the reference amplitude by more than the respective threshold only once.
• the criterion is only met if, for a first set of frequency bands, the respective amplitude is more than the third limit value below the reference amplitude.
• the criterion is only met if, for a second set of frequency bands, the respective amplitude is more than the fourth limit value above the respective reference amplitude.
• the reference amplitude is specifically assigned to the respective frequency band and differs between the individual frequency bands. Consequently, each frequency band is not considered separately, but rather as a whole.
• the first set of frequency bands is greater than one, and the second set is expediently greater than two. Therefore, the matching process only occurs if at least a slight comb-like structure is present in the frequency spectrum, comprising at least three peaks and two notches. This prevents the desired effect or similar from being hindered by the matching process, thus improving usability.
• the criterion is only met if the ambient noise is assigned to a specific class. In other words, the ambient noise is first examined and classified. A comparatively large number of stationary noises is considered a key criterion for classification. This is particularly the case when there are a relatively large number of speakers, engine noise predominates, especially traffic noise, or when machinery is in motion.
• the criterion is only met if the strength of the comb filter artifact exceeds a certain threshold, such as 0.5, or if the comb filter artifact occurs as a result of the corresponding processing.
• the criterion is not met, in particular regardless of whether any other conditions are met, if the frequency band at which the ratio between the upper and lower limit lies is used only for speech.
• This criterion is not met if the frequency band corresponds to, in particular, the user's speech or the speech of other people.
• the amplitude in this frequency band is solely caused by speech, and no other sound sources for these frequencies are present.
• the criterion is not met if the frequency band corresponds only to tonal music. In this case, the corresponding amplitude is present due to music that the user wishes to perceive, so that a change in the amplitude would lead to a distorted musical experience.
• the presence of a comb filter artifact is rather unlikely in this case, or at least it would not be perceived as disturbing.
• a hearing aid can be, for example, a headset or, more commonly, a hearing aid device. Examples include receiver-in-the-canal (RIC) hearing aids, in-the-ear (ITC) hearing aids, complete-in-canal (CIC) hearing aids, hearing glasses, or pocket hearing aids. Alternatively, a hearing aid can be a behind-the-ear (BTE) hearing aid, which is worn behind the ear.
• RIC receiver-in-the-canal
• ITC in-the-ear
• CIC complete-in-canal
• hearing aid can be a behind-the-ear (BTE) hearing aid, which is worn behind the ear.
• BTE behind-the-ear
• the hearing aid includes a microphone.
• This microphone is, for example, omnidirectional, or its directional characteristics can be adjusted.
• the microphone preferably has two or more microphone units.
• the microphone is designed and configured to capture ambient sound.
• an input signal is generated by the microphone when ambient sound is detected.
• the hearing aid advantageously includes a signal processing unit, which is preferably connected to the microphone.
• the input signal is fed to the signal processing unit during operation.
• the hearing aid includes a receiver, which outputs a signal that advantageously corresponds to the processed audio signal and is advantageously connected to the signal processing unit.
• the hearing aid operates according to a method in which the microphone generates an input signal based on ambient sound. An output signal is provided based on this input signal, and an output sound is emitted via the receiver based on this output signal.
• the unprocessed ambient sound present in the receiver's vicinity is determined, and the ratio between the unprocessed ambient sound and the output sound is calculated for different frequency bands.
• the amplitude of the output signal for the frequency band is then adjusted to match the ratio between an upper and a lower limit.
• the signal processing unit is suitable, and in particular designed and configured, to perform at least one or both of these methods.
• the invention further relates to a hearing aid system with two such hearing aids, i.e., a binaural hearing aid system.
• the method is carried out separately using each hearing aid.
• a signal exchange takes place between them, and if the amplitude is changed in one of the hearing aids, the amplitude is also changed in the other hearing aid at the same frequency band, expediently regardless of the ratio between the two.
• coordinated behavior is maintained, which does not lead to a deterioration of the acoustic experience for the user.
• FIG 1 A simplified schematic representation of a hearing aid 2 is shown.
• the hearing aid 2 has a housing 4, inside which a microphone 6 is arranged. Ambient sound 7 can be detected by means of the microphone 6.
• the microphone 6 has several microphone units (not shown in detail), each designed as an electromechanical transducer or a capacitive transducer.
• a signal processing unit 8 is connected downstream of the microphone 6.
• a receiver 10 is connected downstream of the signal processing unit 8, by means of which, when used as intended by a user, it is possible to output sound 12 into the ear canal of the user (not shown in detail).
• the signal processing unit 8 includes a processing unit 14, which processes an input signal 16 provided by the microphone 6 during operation, resulting in an output signal 18. This process begins with splitting the signal into different frequency bands, with each band being processed separately. This allows for frequency-selective amplification and/or attenuation, suppressing noise or other unwanted sounds.
• the processing unit 14 includes, for example, a digital sound processor. Compression is also performed, reducing the frequency spectrum of the output signal 18 compared to the input signal 16.
• the output signal 18 is routed to the receiver 10, so that the output sound 12 emitted by means of the receiver 10 corresponds to the output signal 18.
• the signal processing unit 8 also includes an estimation unit 20, which also receives the input signal 16.
• the estimation unit 20 comprises two transfer functions. One of these determines the resulting output sound 12 based on the input signal 16. This is done theoretically, for which the operations to be performed by the processing unit 14 are simulated and a frequency response of the receiver 10 is taken into account.
• the other transfer function estimates the portion of the ambient sound 7 entering the ear canal past the hearing aid 2, so that the unprocessed ambient sound 23 present in the area of the receiver is determined.
• the geometric design of the housing 4, such as holes, and/or the ear canal is taken into account. This transfer function is created based on a measurement and is adapted to the respective user.
• the processing unit 14 and/or the receiver 10 are not operated, i.e., if the output sound 12 is not generated, the user thus only perceives the unprocessed ambient sound 22, which is dampened, for example, due to the presence of the hearing aid 2 compared to the ambient sound 7.
• FIG. 2 A procedure 24 for operating the hearing aid 2 is shown.
• the input signal 16 is generated using the microphone 6 based on the ambient sound 7. This signal is then sent to the signal processing unit 8, namely to the processing unit 14 and the estimation unit 20.
• the output sound 12 and the unprocessed ambient sound 22 are determined using the estimation unit 20 based on the transfer functions. It is possible that, due to the superposition of the output sound 12 with the unprocessed ambient sound 22 in the ear canal, particularly in the area of the eardrum, a frequency spectrum 30 shown in Figure 3 results. In the frequency spectrum 30 shown, a comb filter artifact 32 occurs due to unfavorable interference between the output sound 12 and the unprocessed ambient sound 22. Several comparatively sharp minima are present, which have a predetermined frequency spacing from each other. These depend on the frequency-specific amplification as well as the anatomy of the ear canal and therefore differs between different users.
• a ratio 36 between the unprocessed ambient sound 22 and the output sound 12 is determined for the different frequency bands into which the input signal 16 is divided.
• the respective ratio 36 is determined only for all frequency bands with frequencies between 250 Hz and 2 kHz.
• a fourth processing step 42 is performed.
• the upper limit 38 is set at 6 dB and the lower limit 40 at -6 dB.
• the fourth processing step 42 the input signal 16 is processed by the processing unit 14 according to the user's potential hearing loss, and the output signal 18 is generated based on the input signal 16.
• the output signal 18 is then routed to the receiver 10, so that the output sound 12 is emitted by the receiver 10 based on the output signal 18.
• a fifth step 44 is performed. In this step, it is checked whether a criterion 46, which comprises several conditions, is fulfilled. For example, fulfilling one condition is sufficient for criterion 46 to be fulfilled. Alternatively, fulfilling further conditions is required for criterion 46 to be fulfilled.
• the ambient noise is classified as 7. This involves checking for the presence of stationary noise caused by speech from many speakers. It also checks whether the stationary noise is caused by engine noise, such as car noise or the operation of heavy machinery. If this is the case, then criterion 46 is met. Consequently, criterion 46 is only met if the ambient noise is assigned to a specific class.
• criterion 46 is also fulfilled.
• the respective amplitude is smaller than the respective reference amplitude by more than the third limit. It is also checked whether, for several of the frequency bands (specifically, for a second number of 3), the respective amplitude is larger than the respective reference amplitude by more than the fourth limit. If this condition is met, the comb filter artifact 32 is present.
• criterion 46 is always not met if the frequency band corresponds solely to speech or tonal music. That is, if only speech or tonal music contributes to the respective amplitude in the respective frequency band.
• the speech can be produced by a single other speaker or by the user of hearing aid 2 themselves.
• criterion 46 is not met, for example, because none, not all, or at least not a predetermined number of conditions are fulfilled, or if the frequency band corresponds solely to speech or tonal music. Otherwise, a sixth processing step 58 is carried out. In this step, a gain factor 50 assigned to the frequency band is changed, namely reduced. This is done for all frequency bands where the ratio 36 between the two limit values 38 and 40 is correct, and where criterion 46 is met. The changed gain factors 50 are then passed to processing unit 14.
• the fourth step, 42 is then carried out again.
• the amplification factors 50 modified in the sixth step 48, are used for the respective frequency bands.
• the amplification factors specified based on the hearing loss are used.
• the amplitude of the output signal 18 is altered. Consequently, the resulting ratio of the frequency bands between the unprocessed ambient sound 22 and the output sound 12 is either above the upper limit 38 or below the lower limit 40, thus avoiding the formation of the comb filter artifact 32. Since the ratio 36 is only created for frequency bands between 250 Hz and 200 kHz, only the amplitude for the frequency bands with frequencies between 250 Hz and 2 kHz is changed. Furthermore, the amplitude of the frequency bands is only changed if criterion 46 is met.
• the changes to the amplification factor 50 in the sixth work step 48 depend on the ratio 36.
• the graph shows a function of the changes in the gain factor 50 against the ratio 36, with dB Id used as the unit on the axes.
• dB Id used as the unit on the axes.
• 10 dB Id is subtracted from the gain factor to account for the change, resulting in a reduction of 60 dB.
• the reduction decreases linearly.
• 5 dB Id is subtracted from the gain factor 50. Consequently, the function is essentially notched.
• the function specifies the target value for the change in the amplification factor of 50.
• the change itself occurs gradually, over a specific timeframe, towards the target value specified by the function. This avoids a harsh transition that would otherwise be perceived as unpleasant by the user.

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