Method for operating a hearing aid system

The hearing aid system addresses the challenge of uncomfortable speech volumes by separating and processing ambient sound components with tailored amplification, enhancing user comfort and conversation clarity through dynamic speech volume adjustment.

EP4615005A1Pending Publication Date: 2025-09-10SIVANTOS PTE LTD
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Patent Information

Application Number
EP2025157453
Authority / Receiving Office
EP · EP
Patent Type
Applications
Current Assignee / Owner
Priority Date
2024-03-07
Filing Date
2025-02-12
Publication Date
2025-09-10

AI Technical Summary

Technical Problem

Hearing aid users often speak more quietly due to sound amplification, making it difficult for others to understand them, or they speak at a volume uncomfortable for others due to strong amplification, leading to unpleasant conversations.

Method used

A hearing aid system that separates ambient sound into components, processes them with different amplification factors based on the user's speech and environmental noise, adjusting the user's speech volume to improve intelligibility and comfort through the Lombard effect.

Benefits of technology

Enhances user comfort and improves conversation clarity by dynamically adjusting speech volume based on environmental noise, ensuring the user is heard clearly without excessive effort.

✦ Generated by Eureka AI based on patent content.

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Abstract

The invention relates to a method (18) for operating a hearing aid system (2), in which an input signal (24) is generated based on ambient sound (22). The input signal (24) is divided into a first signal component (36) and a second signal component (38), wherein the first signal component (36) corresponds to speech (26) of a user and the second signal component (38) does not correspond to speech (26) of the user. A first processed signal (52) is generated based on the first signal component (36) and a first amplification factor (46), and a second processed signal (66) is generated based on the second signal component (38) and a second amplification factor (62). The two processed signals (52, 66) are combined to form an output signal (70). The first amplification factor (46) is selected as a function of a ratio of the first signal component (36) to the second signal component (38).Furthermore, the invention relates to a hearing aid system (2) and a method (74) for commissioning a hearing aid system (2).
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Description

[0001] The invention relates to a method for operating a hearing aid system. Furthermore, the invention relates to a hearing aid and a method for commissioning such a hearing aid system.

[0002] People suffering from hearing loss typically use a hearing aid. This device typically uses an electromechanical transducer to detect ambient sound. The electrical signals generated from the ambient sound are amplified by an amplifier circuit and fed into the person's ear canal via another electromechanical transducer in the form of a receiver. The detected sound signals are usually also processed, for which a signal processor in the amplifier circuit is typically used. The amplification is adjusted to any hearing loss of the hearing aid wearer, who is also referred to as the user or wearer.

[0003] When the user speaks, this is also recorded by the electromechanical transducer, amplified according to the selected gain, and then fed into the ear canal. As a result, the user perceives their own speech as louder than it actually is. This leads to the user tending to speak more quietly. This, in turn, makes it difficult for the other person to follow the user. However, if the user continues to speak at a volume that is comfortable for the other person, this becomes unpleasant for the user due to the comparatively strong amplification.

[0004] The invention is based on the object of specifying a particularly suitable method for operating a hearing aid system as well as a particularly suitable hearing aid system and a particularly suitable method for commissioning a hearing aid system, wherein in particular comfort for a user is increased and / or conversation is improved.

[0005] With regard to the method for operating a hearing aid system, this object is achieved according to the invention by the features of claim 1, with regard to the hearing aid system by the features of claim 7, and with regard to the method for commissioning a hearing aid system by the features of claim 8. Advantageous further developments and refinements are the subject of the respective subclaims.

[0006] The method serves to operate a hearing aid system. The hearing aid system comprises a hearing aid. For example, the hearing aid is a pair of headphones or comprises headphones, and the hearing aid is, for example, a headset. However, the hearing aid is particularly preferably a hearing aid. The hearing aid serves to support a person suffering from a reduced hearing ability. In other words, the hearing aid is a medical device used to compensate for, for example, a partial hearing loss. The hearing aid is, for example, a receiver-in-the-canal hearing aid (RIC), an in-the-ear hearing aid, such as an in-the-ear hearing aid, an in-the-canal hearing aid (ITC), or a completely-in-canal hearing aid (CIC), a pair of hearing glasses, or a pocket hearing aid. Alternatively, the hearing aid is a behind-the-ear hearing aid that is worn behind one ear.

[0007] The hearing aid is intended and configured to be worn on the human body. In other words, the hearing aid preferably comprises a holding device by means of which it can be attached to the human body. If the hearing aid is a hearing aid device, the hearing aid is intended and configured to be placed, for example, behind the ear or within an ear canal. In particular, the hearing aid is wireless and intended and configured to be inserted at least partially into an ear canal.

[0008] The hearing aid preferably comprises a microphone used to capture sound. In particular, during operation, ambient sound, i.e., sound waves, is captured by the microphone, or at least a portion thereof. The microphone is expediently arranged at least partially within a housing of the hearing aid and thus at least partially protected. The microphone is suitably an electromechanical sound transducer. The microphone has, for example, only a single microphone unit or several microphone units that interact with one another. Each of the microphone units expediently has a membrane that is set into vibration by sound waves, wherein the vibrations are converted into an electrical signal by means of a corresponding recording device, such as a magnet that is moved in a coil.Alternatively, the microphone units are designed capacitively, exploiting the fact that an applied electrical voltage changes when the distance between the diaphragm and a static surface of the microphone unit changes. The electrical voltage is applied in particular between the diaphragm and the static surface. The microphone units are preferably designed omnidirectionally. In this or another way, it is at least possible to use the microphone to generate or at least provide an input signal based on the sound impinging on the microphone, namely, in particular, the ambient sound.

[0009] The hearing aid expediently has a receiver for outputting an output signal. The output signal is in particular an electrical signal, and for example, digital or suitably analog. The receiver is preferably an electromechanical sound transducer, for example a loudspeaker. Depending on the design of the hearing aid, in the intended state the receiver is at least partially arranged within an ear canal of a user of the hearing aid, i.e. a person who is also referred to as the wearer, user or hearing aid wearer, or is at least acoustically connected to this. The hearing aid serves in particular primarily to output the output signal via the receiver, whereby a corresponding sound is created. In other words, the main function of the hearing aid is preferably to output the output signal.

[0010] The hearing aid suitably comprises a signal processing unit by means of which the microphone, if present, and the receiver, if present, are connected in terms of signal technology. The hearing aid expediently has a signal processor, which, 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 is implemented using analog components. The signal processor or at least the signal processing unit is used, in particular, to adapt the input signal generated by the microphone. The signal processing unit is at least suitable, in particular provided and configured, for this purpose. An A / D converter is expediently 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.Particularly preferably, the hearing aid additionally comprises 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.

[0011] The hearing aid system is formed, for example, solely by means of the hearing aid. However, the hearing aid system is particularly preferably formed from two such hearing aids, which are in particular structurally identical to one another. Here, one of the hearing aids is assigned to the user's left ear and the other hearing aid to the user's right ear. Consequently, the hearing aid system is designed as a binaural hearing aid system. In a further alternative, the hearing aid system comprises, for example, another device that is expediently portable. The additional device is, for example, a smartphone or other wearable. In particular, the individual components of the hearing aid system are connected to one another via signal technology, for example wired or preferably wirelessly. Expediently, each of the devices of the hearing aid system has a microphone by means of which the ambient sound can be detected.

[0012] The method provides for the input signal to be generated based on the ambient sound. In other words, the ambient sound is captured, and the input signal is generated based on this. The input signal is suitably an electrical signal, and the generation is expediently carried out using the microphone(s). The input signal corresponds, for example, to the unprocessed ambient sound or has already been processed. The input signal expediently has a specific directional characteristic so that a specific part of the environment is captured with amplified frequency, i.e., in particular, sound from a specific spatial angle.

[0013] The input signal is split into a first signal component and a second signal component. For example, the input signal may comprise further components that are assigned to neither the first nor the second signal component. However, it is particularly preferred that the input signal be completely split between the first signal component and the second signal component, so that no further components are present. The first signal component corresponds to the user's speech, whereas the second signal component does not correspond to the user's speech. Thus, the part of the ambient sound that is caused by the user speaking is assigned to the first signal component. The other components, however, are assigned to the second signal component. In particular, the second signal component does not contain any portion corresponding to sound that was caused by the user speaking.For the separation, a spatial analysis is performed, for example, to determine where the ambient sound originated. Alternatively, the separation is performed using frequency analysis or another method. In this case, it is possible that at least one of the signal components is missing, at least temporarily. If the user is silent, the first signal component, in particular, is missing.

[0014] Based on the first signal component and a first amplification factor, a first processed signal is created. Here, for example, the first signal component is amplified using the first amplification factor, so that the first processed signal is created. The first amplification factor is, for example, a constant value. Alternatively, the first amplification factor is, for example, not constant and is in particular dependent on a frequency of the respective individual components of the first signal component. In particular, the first amplification factor relates to amplification, compression and / or directionality. Alternatively or in combination with this, the first amplification factor relates to noise suppression. At least the first signal component is processed using the first amplification factor, so that the first processed signal is created.In other words, the first gain factor suitably corresponds to a parameter set by means of which the first signal component is processed, thereby creating the first processed signal. Preferably, the processing is performed solely by means of the first gain factor, or, for example, further processing steps are performed to create the first processed signal.

[0015] Furthermore, a second processed signal is created based on the second signal component and a second amplification factor. The second amplification factor is, for example, simply a constant value. Alternatively, it depends on a frequency of the individual components of the second signal component. Alternatively, the second amplification factor describes a compression, a directionality and / or a noise suppression setting. At least the second signal component is processed using the second amplification factor in such a way that the second processed signal is created. For example, the second processed signal is created solely as a result of processing using the second amplification factor, or further processing steps are carried out for this purpose.

[0016] In a further step, the two processed signals are combined to form the output signal. In particular, the two processed signals are added together or combined in some other way, for example, by weighted addition. The first signal component, the second signal component, the input signal, and the output signal are, in particular, electrical signals. The corresponding processing is expediently carried out by means of the signal processing unit, if any, suitably the digital signal processor. The output signal is expediently output, for example, via the earpiece, if any, so that output sound is created, which is suitably introduced into the user's ear canal.

[0017] The first and second amplification factors are, for example, always positive, negative, or can, for example, be both negative and positive, expediently depending on specific requirements. The second amplification factor is preferably predetermined depending on any hearing loss of the user. Alternatively or in combination with this, the second amplification factor is predetermined by the user or, in particular, adapted to the user. Preferably, the second amplification factor is selected depending on the ambient sound and / or a classification of the environment. The first amplification factor, on the other hand, is selected depending on a ratio of the first signal component to the second signal component. In particular, the first amplification factor depends on the ratio of the level of the first signal component to the level of the second signal component.

[0018] Due to this process, the user perceives the sound resulting from their speech in a different way. When the user is in a relatively quiet environment, the ratio of the first signal component to the second signal component is different than when the benefit is perceived in a relatively noisy environment. As a result, a different first amplification factor is selected. This makes it possible to exploit the Lombard effect, which describes the fact that people in a relatively noisy environment also (unconsciously) speak louder.

[0019] The first amplification factor is expediently reduced when the ratio of the first signal component to the second signal component is comparatively small and / or below a certain limit, i.e. when the user speaks comparatively quietly compared to the environment. The user therefore perceives themselves as being comparatively quiet, which is why they subsequently speak louder (particularly unconsciously). As a result, the user can be reliably heard by conversation partners even in a comparatively noisy environment. In summary, when the ratio is comparatively small, the first amplification factor is reduced so that the user's own speech is comparatively weak. As a result, the user will speak louder, making it easier for conversation partners to understand the user.

[0020] In contrast, if the ratio of the first signal component to the second signal component is comparatively large, in particular greater than a certain limit, the first amplification factor is preferably increased so that the user can perceive their own speech as comparatively loud. As a result, the user will speak more quietly, so that the conversation is more pleasant for the conversation partner. Because of the quieter speech, despite the increased first amplification factor, the first signal component is not excessively present in the output signal, so that comfort for the user is not reduced. In summary, conversations with conversation partners are improved without overexertion for the user, so that comfort for them is increased.

[0021] For example, a manufacturer of the hearing aid system, for example in the signal processing unit, stores how the first amplification factor is selected depending on the ratio, i.e., in particular, its dependency. Alternatively, or in combination with this, the dependency is adjusted, for example, by a specialist. In another alternative, the dependency is determined using a method for commissioning the hearing aid system. This occurs, in particular, when the hearing aid system is used by the user, i.e., expediently after the hearing aid system has been handed over to the user.

[0022] For example, the input signal is split into only the first and second signal components. However, the second signal component is particularly preferably split into a third signal component and a fourth signal component. This splitting occurs, for example, after prior processing of the second signal component, or expediently, the splitting occurs in a single step with the splitting of the input signal into the first signal component.

[0023] The third signal component corresponds to a desired sound source, in particular sound produced by a conversation partner or the like. The third signal component expediently corresponds to (ambient) sound from a specific spatial area, into which, for example, a directional lobe of the microphone is directed. In contrast, the fourth signal component corresponds to a noise source. A third processed signal is created using the third signal component and a third amplification factor, and a fourth processed signal is created using the fourth signal component and a fourth amplification factor. The third and / or fourth amplification factors are components of the second amplification factor and are configured, for example, according to the first amplification factor.For example, the third and / or fourth amplification factor are each a value that is constant or dependent on certain frequencies of the respective signal component, and / or a certain set of parameters by means of which a compression or the like is set.

[0024] The third processed signal and the fourth processed signal are combined to form the second processed signal. The combination of the third and fourth processed signals preferably takes place in the same work step in which the combination with the first processed signal takes place, so that the second processed signal is created in particular only implicitly. Here, too, it is possible for one of the individual signal components to be absent, at least temporarily, for example the fourth signal component, if there is no noise source. Due to the division into the third and fourth signal components, it is possible to present noises, in particular sound, that are of no interest to the user with a lower amplification, so that the user is not distracted by the noise source. This further increases comfort.

[0025] For example, the first amplification factor is selected depending on the ratio of the first signal component to a combination of the third and fourth signal components. However, the first amplification factor is particularly preferably selected solely, or at least also, depending on the ratio of the first signal component to the fourth signal component. Thus, the user will adjust their speech volume depending on the background noise, in particular the noise source. Consequently, in a comparatively noisy environment, the user will speak louder due to the reduced first amplification factor.

[0026] For example, the first amplification factor is selected solely as a function of the ratio of the first signal component to the fourth signal component. However, the first amplification factor is particularly preferably selected solely, or preferably additionally, as a function of the ratio of the third signal component to the fourth signal component, i.e., in particular, the relative levels of the individual signal components. If the desired sound source corresponds to a conversation partner, the Lombard effect also applies to this partner. If the conversation partner only understands the user relatively poorly, the conversation partner will speak louder, so that the ratio of the third signal component to the fourth signal component is increased.In this case, at least if the ratio is above a certain threshold, the first gain factor is reduced, especially compared to the choice resulting from the ratio of the first signal component to the fourth signal component. Thus, the user subsequently speaks louder, allowing the other party to understand the user better.

[0027] By selecting the first amplification factor, which also depends on the ratio of the third signal component to the fourth signal component, it is monitored whether, for example, the first amplification factor is initially sufficient for the conversation partner to understand the user. In this case, the ratio of the third signal component to the fourth signal component is reduced and corresponds in particular to a certain expected value. If, on the other hand, the ratio remains above a certain limit, the user is comparatively difficult to understand for the conversation partner. In this case, the first amplification factor is reduced in particular so that the user speaks louder and thus intelligibility for the conversation partner is improved.

[0028] For example, the first amplification factor is independent of the noise source. However, it is particularly preferred that the noise source be categorized, for which the fourth signal component in particular is analyzed. In other words, the noise source be assigned to one of several specific categories. For example, it is checked whether the noise source is due to the operation of a machine, or whether the noise source corresponds to a conversation between several other people that the user does not want to follow. Alternatively, or in combination, it is checked whether the noise source corresponds to driving noise due to the use of a vehicle. The first amplification factor is expediently selected depending on the categorization.In this way, particular attention is paid to ensuring that the changing volume of the user's speech does not disturb other people, and / or that the volume is not unpleasant for the conversation partner in the same vehicle. It is always ensured that the (unconscious) change in the user's volume improves the intelligibility of the conversation partner.

[0029] For example, the gain factor is only selected once, particularly if a specific environment is present. In other words, a check is made to see whether the environment has changed, and if there is a change, the first gain factor is selected accordingly. If, on the other hand, the environment does not change, the first gain factor is not adjusted any further. However, it is particularly preferred that the first gain factor is adjusted continuously. This occurs, for example, continuously over time or in certain discrete time intervals, for example every second, every 5 seconds, or every 10 seconds. In this way, a check is carried out to see whether any change / adjustment of the first gain factor leads to a changed ratio of the first signal component to the second signal component. If this corresponds to a specific expected value, the first gain factor is expediently continued to be used and not changed.If, however, the ratio differs from the expected value by more than a certain value, the first gain factor is (further) adjusted so that the user also changes his speaking volume.

[0030] The hearing aid system comprises a hearing aid. The hearing aid is, for example, a headset or, particularly preferably, a hearing aid. For example, the hearing aid is a receiver-in-the-canal (RIC) hearing aid, an in-the-ear hearing aid, such as an in-the-ear (ITC) hearing aid, or a completely in-canal (CIC) hearing aid, a pair of hearing glasses, or a pocket hearing aid. Alternatively, the hearing aid is a behind-the-ear (behind-the-ear) hearing aid that is worn behind an auricle. For example, the hearing aid system is formed by the hearing aid or comprises at least one other device, such as another hearing aid that is, in particular, structurally identical. Alternatively, the other device is another device that is, in particular, portable, such as a smartphone.

[0031] The hearing aid system has a microphone. This is, for example, omnidirectional, or suitably, it is possible to change the directional characteristic of the microphone. For this purpose, the microphone preferably has two or more microphone units. The microphone is suitable, in particular provided and configured, for detecting ambient sound. An input signal is expediently generated by the microphone when the ambient sound is detected. The hearing aid system, in particular the hearing aid, further has a signal processing unit, which is preferably connected to the microphone for signal processing purposes. In particular, the input signal is fed to the signal processing unit during operation.

[0032] The hearing aid system is operated according to a method in which the input signal is created based on the ambient sound. The input signal is split into a first signal component and a second signal component, wherein the first signal component corresponds to a user's speech and the second signal component does not correspond to the user's speech. A first processed signal is created based on the first signal component and a first amplification factor, and a second processed signal is created based on the second signal component and a second amplification factor. The two processed signals are combined to form an output signal. The first amplification factor is selected depending on a ratio of the first signal component to the second signal component. Expediently, the signal processing unit is suitable, in particular provided and configured, to at least partially carry out the method.

[0033] The procedure for commissioning the hearing aid system is performed before the procedure for operating the hearing aid system is performed. The procedure for commissioning the hearing aid system is used to determine the dependency of the first amplification factor. For example, the procedure for commissioning the hearing aid system is repeated several times, particularly at periodic intervals. Alternatively, the procedure is performed only once or after initialization by a user.

[0034] During the commissioning process, the input signal is generated based on ambient sound. The microphone is also expediently used for this purpose. Furthermore, the input signal is split into the first signal component and the second signal component, with the first signal component corresponding to the user's speech and the second signal component not corresponding to the user's speech. The input signal is expediently generated and / or split into the two signal components in the same way as during subsequent operation of the hearing aid according to the (other) method.

[0035] After splitting the signal components, the ratio of the first signal component to the second signal component is determined. To do this, the respective input signal is expediently created in various different environments based on the respective ambient sound, this signal is split into the signal components, and the ratio is determined. The first gain factor is expediently determined using linear regression so that a desired ratio is formed. In particular, the ratio is formed for each environment, for which the respective levels are used in particular. After this has been done several times, the linear regression is carried out. To do this, a histogram is expediently first created and the median of the gradient as well as the offset of the associated straight line are determined. The gradient then corresponds to the dependence of the first gain factor on the ratio.Preferably, the gradient of the straight line is limited to a value between 0.3 dB and 0.7 dB.

[0036] Suitably, the first amplification factor is constant above a certain ratio, so that excessive change in the first signal component is prevented. Suitably, the first amplification factor or at least the change in the first amplification factor corresponds to the minimum of 0 or any value and the quotient of a difference and an auxiliary value. To calculate the difference, the expected value is subtracted from the ratio of the two signal components. The auxiliary value is preferably adapted to the user and is preferably between 0.3 dB and 0.7 dB. The auxiliary value is preferably determined using linear regression.

[0037] Due to the procedure, the initial amplification factor is selected appropriately for different users, as the Lombard effect differs from person to person. Consequently, no adjustment by the manufacturer and / or a specialist, such as an audiologist, is required, while the procedure is still relatively precisely tailored to the user.

[0038] The further developments and advantages explained in connection with the two procedures can also be transferred to the hearing aid system and to each other, and vice versa.

[0039] An embodiment of the invention is explained in more detail below with reference to a drawing. In the drawings: Fig. 1 schematically simplified a hearing aid system, Fig. 2 a method for operating the hearing aid system, Fig. 3 a curve of a first amplification factor, and Fig. 4 a method for commissioning the hearing aid system.

[0040] Corresponding parts are provided with the same reference numerals in all figures.

[0041] In Figure 1A hearing aid system 2 is shown in a simplified schematic detail, comprising a hearing aid 4. The hearing aid 4 has a housing 6, within which a microphone 8 is arranged. The microphone 8 has a plurality of microphone units (not shown in detail), each of which is designed as an electromechanical sound transducer or a capacitive sound transducer. A signal processing unit 10, which has a control unit 12, is connected downstream of the microphone 8. A receiver 14 is connected downstream of the signal processing unit 10, by means of which receiver it is possible, when used as intended by a user, to output sound into an ear canal of the user (not shown in detail). In addition, the hearing aid 4 has a communication device 16, which is also connected to the signal processing unit 10.

[0042] The hearing aid system 2 comprises two such hearing aids 4, which are signal-linked to each other during use via their respective communication devices 16. A Bluetooth standard is used for the signal-linking connection. One of the hearing aids 4 is assigned to the user's left ear and the other to the user's right ear, so that the hearing aid system 2 is designed binaurally.

[0043] In Figure 21 shows a method 18 for operating the hearing aid system 2, which is carried out at least partially by means of the signal processing unit 10 of each hearing aid 4. In a first work step 20, an input signal 24 is created based on ambient sound 22. For this purpose, the ambient sound 22 impinging on the respective microphone 8 from outside the housing 6 is recorded by each of the microphones 8 and converted into the electrical input signal 24, which is passed to the signal processing unit 10. The ambient sound 22 is composed of three components, one of which represents the speech 26 of the user themselves. Another component of the ambient sound 22 is present due to a conversation partner and is thus present due to a desired sound source 28. The third component is caused by an interference noise source 30 and is of no interest to the user.

[0044] In a subsequent second work step 32, the input signal 24 is split into a first signal component 36 and a second signal component 38 by means of a splitting unit 34 of the signal processing unit 10. The second signal component 38 is composed of a third signal component 40 and a fourth signal component 42, so that the input signal 24 is split directly into the first signal component 36, the third signal component 40, and the fourth signal component 42 by means of the splitting unit 34.

[0045] The first signal component 36 corresponds to the user's speech 26, and the second signal component 38 does not correspond to the user's speech. In other words, the first signal component 36 contains only the speech 26 of the ambient sound 22. The second component 38, however, comprises the remaining part of the ambient sound 22. The third signal component 40 corresponds to the desired sound source 28, whereas the fourth signal component 42 corresponds to the noise source 30. In other words, the third signal component 40 designates the part of the ambient sound 22 that was caused by the desired sound source 28, whereas the fourth signal component 42 corresponds to the portion of the ambient sound 22 that is present solely due to the noise source 30. For the division, a spatial analysis is used to check where the individual components of the ambient sound 22 originate.For this purpose, the directional characteristics of the microphones 8 of the two hearing aids 4 are adjusted / checked to each other, for which the communication devices 16 of the two hearing aids 4 are used.

[0046] The noise sources 30 are also categorized, for which the fourth signal component 42 is checked. This check determines whether the noise source 30 corresponds to a conversation between other people, or whether the noise emitted by the noise source 30 is wind noise from a vehicle or is caused by the operation of a machine.

[0047] In a subsequent third step 44, a first amplification factor 46 is selected. This is dependent on a ratio 48 of the first signal component 36 to the fourth signal component 42, namely, the level of the first signal component 36 to the level of the fourth signal component 42. Thus, the first amplification factor 46 is dependent on the ratio of the first signal component 36 to the second signal component 38, namely, the fourth signal component 42 to the second signal component 38.

[0048] In Figure 3The dependence of the first amplification factor 46 on the ratio 48 is shown, with the level of amplification indicated on the ordinate. The first amplification factor 46 is determined using a formula, namely the minimum of a specific / specified value, such as 0, and a quotient. The quotient is formed from a difference and an auxiliary value that lies between 0.3 dB and 0.6 dB and is adapted to the user. To form the difference, an expected value is subtracted from the current ratio 48. Thus, the first amplification factor 46 has an at least partial / section-wise linear dependence on the ratio 48. In addition, the categorization of the noise source 30 is taken into account when selecting the first amplification factor 46. Depending on the category, a different auxiliary value is used, so that the gradient is different, as can also be seen in the graph in Figure 3In summary, the first amplification factor 46 is also selected depending on the categorization of the noise source 30.

[0049] In addition, the first gain factor 46 is also selected depending on the ratio of the third signal component 40 to the fourth signal component 42, which also has a linear relationship. A correspondingly adapted formula is used for this purpose.

[0050] After the first amplification factor 46 has been determined, a fourth work step 50 is performed. In this step, the first signal component 36 is processed using the first amplification factor 46, so that a first processed signal 52 is created. For this purpose, in particular, the first signal component 36 is multiplied by the first amplification factor 46. In addition, the third signal component 40 is processed using a third amplification factor 54, so that a third processed signal 56 is created. The third amplification factor 54 is adapted to the user's hearing loss. The fourth signal component 42 is processed using a fourth amplification factor 58, so that a fourth processed signal 60 is created.The fourth amplification factor 58 is comparatively small, so that the ratio of the third processed signal 56 to the fourth processed signal 60 is increased compared to the ratio of the third signal component 40 to the fourth signal component 42.

[0051] In summary, the third processed signal 56 is created using the third signal component 40 and the third gain factor 54, and the fourth processed signal 60 is created using the fourth signal component 42 and the fourth gain factor 58. The third gain factor 54 and the fourth gain factor 58 together form a second gain factor 62, by means of which the second signal component 38 is processed.

[0052] In a subsequent fifth step 66, the third processed signal 56 and the fourth processed signal 60 are combined, namely added, to form a second processed signal 64 by means of an adder 65 of the signal processing unit 10. At the same time, the second processed signal 64 thus created is combined with the first processed signal 52 to form an output signal 70, which is output by the adder 65.

[0053] In a subsequent sixth step 72, the output signal 70 is output via the handset 14 and thus presented to the user. Due to the corresponding selection of the first amplification factor 46, the user's own speech is presented to the user at a modified volume, so that the user adjusts their speech volume due to the Lombard effect. The hearing aid system 2 thus takes the Lombard effect into account / exploited, so that the user of the hearing aid system 2 adjusts their speech volume (speech volume) such that the ratio of the first signal component 36 to the fourth signal component 42, i.e., their speech volume to the volume of the noise source 30, reaches a specific expected value. This increases intelligibility for the user's conversation partner. The method 18 is carried out continuously, so that the first amplification factor 46 is continuously adjusted.

[0054] In Figure 4 a method 74 for commissioning the hearing aid system 2 is shown. In a seventh work step 76, the input signal 24 is provided based on the ambient sound 22. The seventh work step 76 is essentially the same as the first work step 20. In a subsequent eighth work step 78, the input signal 24 is divided into the first signal component 36 and the second signal component 38. Here, too, the further division into the third and fourth signal components 40, 42 takes place, and the eighth work step 78 is essentially the same as the second work step 32. Here, too, the possible noise source 30 is categorized. The seventh and eighth work steps 76, 78 are carried out several times in different environments, in particular if different categories of noise sources 30 are present.

[0055] In a subsequent ninth step 80, the first amplification factor 46 for the different categories of noise sources 30 is determined based on the respective ratios 48 of the first signal component 36 to the second signal component 38, namely the Figure 3 The graph shown here is used to determine the predefined expected value, and the respective auxiliary value is determined, particularly for the different categories of noise sources 30. For this purpose, a histogram of the determined ratios 48 is created, and the median is used as the respective auxiliary value.

[0056] The invention is not limited to the exemplary embodiment described above. Rather, other variants of the invention can also be derived therefrom by those skilled in the art without departing from the scope of the invention. In particular, all individual features described in connection with the exemplary embodiment can also be combined with one another in other ways without departing from the scope of the invention. List of reference symbols

[0057] 2 Hearing aid system 4 Hearing aid 6 Housing 8 Microphone 10 Signal processing unit 12 Control unit 14 Receiver 16 Communication device 18 Method for operating a hearing aid system 20 First work step 22 Ambient sound 24 Input signal 26 Speech 28 Desired sound source 30 Noise source 32 Second work step 34 Splitting unit 36 ​​First signal component 38 Second signal component 40 Third signal component 42 Fourth signal component 44 Third work step 46 First gain factor 48 Ratio 50 Fourth work step 52 First processed signal 54 Third gain factor 56 Third processed signal 58 Fourth gain factor 60 Fourth processed signal 62 Second gain factor 64 Fifth work step 65 Adder 66 Second processed signal 70 Output signal 72Sixth step 74Procedure for commissioning a hearing aid system 76Seventh step 78Eighth step 80Ninth step

Claims

1. Method (18) for operating a hearing aid system (2), in which - an input signal (24) is created on the basis of ambient sound (22), - the input signal (24) is divided into a first signal component (36) and a second signal component (38), wherein the first signal component (36) corresponds to speech (26) of a user and the second signal component (38) does not correspond to speech (26) of the user, - a first processed signal (52) is created on the basis of the first signal component (36) and a first amplification factor (46), - a second processed signal (66) is created on the basis of the second signal component (38) and a second amplification factor (62), and - the two processed signals (52, 66) are combined to form an output signal (70), wherein the first amplification factor (46) is selected as a function of a ratio of the first signal component (36) to the second signal component (38).

2. Method (18) according to claim 1,characterized by - that the second signal component (38) is divided into a third signal component (40) and a fourth signal component (42), wherein the third signal component (40) corresponds to a desired sound source (28) and the fourth signal component (42) corresponds to a noise source (30), - that a third processed signal (56) is created based on the third signal component (40) and a third amplification factor (54), - that a fourth processed signal (60) is created based on the fourth signal component (42) and a fourth gain factor (58), and - that the third processed signal (56) and the fourth processed signal (60) are combined to form the second processed signal (66).

3. Method (18) according to claim 2, characterized by thatthe first amplification factor (46) is selected as a function of a ratio (48) of the first signal component (36) to the fourth signal component (42).

4. Method (18) according to claim 2 or 3, characterized by that the first amplification factor (46) is also selected as a function of a ratio of the third signal component (40) to the fourth signal component (42).

5. Method (18) according to one of claims 2 to 4, characterized by that the noise source (30) is categorized, and that the first amplification factor (46) is also selected depending thereon.

6. Method (18) according to one of claims 1 to 5, characterized by that the first gain factor (46) is continuously adjusted.

7. Hearing aid system (2) comprising a hearing aid (4) with a microphone (8) for detecting ambient sound (22) and a signal processing unit (10), and which is operated according to a method (18) according to one of claims 1 to 6.

8. Method (74) for commissioning a hearing aid system (2) according to claim 7, in which - an input signal (24) is created on the basis of an ambient sound (22), - the input signal (24) is divided into a first signal component (36) and a second signal component (38), wherein the first signal component (36) corresponds to speech (26) of a user and the second signal component (38) does not correspond to speech (26) of the user, and - a dependency of the first amplification factor (46) on the ratio of the first signal component (36) to the second signal component (38) is determined.

Citation Information

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