Method for operating a hearing aid

The hearing aid system addresses the Lombard effect by separating user and conversation partner speech and adjusting gain factors to equalize signal levels, enhancing user comfort and conversation clarity.

EP4661433A1Pending Publication Date: 2025-12-10SIVANTOS PTE LTD
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Patent Information

Application Number
EP2025178760
Authority / Receiving Office
EP · EP
Patent Type
Applications
Current Assignee / Owner
Priority Date
2024-06-07
Filing Date
2025-05-26
Publication Date
2025-12-10

AI Technical Summary

Technical Problem

Hearing aid users often experience difficulty understanding their conversation partners due to background noise, leading to the Lombard effect where they unconsciously speak louder, causing discomfort for both parties.

Method used

A hearing aid system that separates user's speech and conversation partner's speech using spatial analysis or frequency methods, applies different gain factors to each signal component, and adjusts the second gain factor to equalize the levels of the processed signals, preventing excessive shifts and improving intelligibility.

Benefits of technology

Enhances user comfort and improves conversation clarity by ensuring the user can understand their conversation partner without needing to speak louder, mitigating the Lombard effect.

✦ Generated by Eureka AI based on patent content.

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Abstract

The invention relates to a method (14) for operating a hearing aid (2) in which an input signal (20) is generated from ambient sound (18). A first signal component (32) and a second signal component (43) are extracted from the input signal (20), wherein the first signal component (32) corresponds to speech (22) of a user and the second signal component (34) corresponds to speech (24) of another person. A first processed signal (42) is generated from the first signal component (32) and a first gain factor (40), and a second processed signal (54) is generated from the second signal component (34) and a second gain factor (52). The two processed signals (42, 54) are combined to form an output signal (80). The second gain factor (52) is selected depending on a difference (48) between the level of the first processed signal (42) and the level of the second processed signal (54).Furthermore, the invention relates to a hearing aid (2).
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Description

[0001] The invention relates to a method for operating a hearing aid and to a hearing aid itself. The hearing aid comprises a microphone for detecting ambient sound and a signal processing unit.

[0002] People with hearing loss typically use a hearing aid. This usually involves an electromechanical transducer that captures ambient sound. The resulting electrical signals are amplified by an amplifier circuit and then delivered to the ear canal via another electromechanical transducer, typically a receiver. The captured sound signals are usually processed, typically by a signal processor within the amplifier circuit. The amplification is adjusted to the specific hearing loss of the hearing aid user. When the user speaks, this is also captured by the electromechanical transducer, amplified according to the selected gain, and delivered to the ear canal.

[0003] To ensure that the sounds of interest to the user are audible in both loud and quiet environments, without excessive and therefore unpleasant amplification, an automatic gain control system is used. This system amplifies the sounds present in the environment according to their specific characteristics, maintaining a level between a predefined minimum and maximum threshold. In other words, the amplification is adjusted to the current environment, so that loud sounds are perceived as loud and quiet sounds as quiet.

[0004] However, it's possible that the user may only perceive their own voice or that of a conversation partner relatively poorly due to background noise. The user's natural reaction to this is to speak louder, thus encouraging the conversation partner to also speak louder. This phenomenon is known as the Lombard effect. However, if automatic amplification is active, the user speaking louder doesn't lead to them perceiving themselves as louder. Consequently, the user will speak even louder, which then results in a loss of comfort for the conversation partner.

[0005] The invention is based on the objective of specifying a particularly suitable method for operating a hearing aid and a particularly suitable hearing aid, wherein in particular comfort for a user is increased and / or conversation is improved.

[0006] With regard to the method, this problem is solved according to the invention by the features of claim 1, with regard to the hearing aid by the features of claim 8, and with regard to the method for commissioning a hearing aid system by the features of claim 8. Advantageous further developments and embodiments are the subject of the respective dependent claims.

[0007] The process is used to operate a hearing aid. For example, the hearing aid is a headphone or includes a headphone, and the hearing aid is, for example, a headset. However, 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. In other words, 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 (ITE) hearing aid, such as an in-the-ear (ITC) hearing aid, a complete-in-the-canal (CIC) hearing aid, hearing glasses, or a pocket hearing aid. Alternatively, the hearing aid is a behind-the-ear (BTE) hearing aid, which is worn behind the ear.

[0008] The hearing aid is designed and configured to be worn on the human body. In other words, the hearing aid preferably includes a retention device that allows it to be attached to the human body. If the hearing aid is a hearing assistance device, it is designed and configured to be placed, for example, behind the ear or within the ear canal. In particular, the hearing aid is wireless and designed and configured to be inserted, at least partially, into the ear canal.

[0009] The hearing aid preferably includes a microphone for capturing sound. In particular, when in operation, the microphone captures ambient sound, i.e., sound waves, or at least a portion thereof. The microphone is advantageously located at least partially within the 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 advantageously has a diaphragm that is set into vibration by sound waves, and the vibrations are converted into an electrical signal by means of a suitable recording device, such as a magnet moved within a coil.Alternatively, the microphone units are designed capacitively, utilizing the fact that an applied electrical voltage changes when the distance between the diaphragm and a static surface of the microphone unit changes. In this case, the electrical voltage is applied specifically 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 generate or at least provide an input signal using the microphone, based on the sound incident on the microphone, namely, in particular, ambient sound.

[0010] Advantageously, the hearing aid includes 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. Depending on the design of the hearing aid, in its intended state, the receiver is at least partially positioned 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. In other words, the main function of the hearing aid is preferably to output the signal.

[0011] The hearing aid suitably includes a signal processing unit by means of which the microphone and receiver are connected. Advantageously, 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. The signal processor, or at least the signal processing unit, is used in particular to adapt the input signal generated by the microphone. At a minimum, the signal processing unit is suitable for this purpose, and in particular, is designed and configured accordingly. Advantageously, 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.Preferably, the hearing aid also 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 in terms of signal processing.

[0012] The method involves generating the input signal from the ambient sound. In other words, 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.

[0013] A first and a second signal component are extracted from the input signal. For example, the input signal may contain other components that are not assigned to either the first or second signal component. The first signal component corresponds to the user's speech, while the second signal component corresponds to the speech of another person. Thus, the portion of the ambient sound generated by the user's speech is assigned to the first signal component. The portion of the ambient sound generated by the other person's speech is assigned to the second signal component. For this assignment, a spatial analysis is performed, for example, to determine the origin of the ambient sound. Alternatively, the division can be carried out using frequency analysis or other methods.

[0014] If the input signal lacks both signal components, particularly for a specific period such as 5 minutes, 2 minutes, 1 minute, 30 seconds, or 10 seconds (as is the case, for example, during a conversation), the procedure is appropriately terminated. The procedure is appropriately initiated only when both the first and second signal components are present in the input signal and / or when a specific operating mode of the hearing aid is selected.

[0015] A first processed signal is created using the first signal component and a first gain factor. For example, the first signal component is amplified by the first gain factor, resulting in the first processed signal. The first gain factor can be a constant value. Alternatively, the first gain factor can be variable and, in particular, dependent on the frequency of the individual components of the first signal component. Specifically, the first gain factor can apply amplification, compression, and / or directionality. Alternatively, or in combination, the first gain factor can apply noise reduction. At a minimum, the first signal component is processed by the first gain factor, resulting in the first processed signal.In other words, the first gain factor suitably corresponds to a set of parameters by which the first signal component is processed, thus creating the first processed signal. Preferably, this processing is carried out solely by means of the first gain factor, or, for example, further processing steps may be performed to create the first processed signal.

[0016] Furthermore, a second processed signal is created using the second signal component and a second gain factor. The second gain factor can be, for example, simply a constant value. Alternatively, it can depend on the frequency of the individual components of the second signal component. Alternatively, the second gain factor can be used to apply compression, directionality, and / or noise reduction. At a minimum, the second gain factor processes the second signal component in such a way as to create the second processed signal. For example, the second processed signal may be created solely as a result of processing with the second gain factor, or further processing steps may be performed.

[0017] In a further step, the two processed signals are combined to form the output signal. Specifically, the two processed signals are added together or combined in some other way, for example, weighted summation. The first signal component, the second signal component, the input signal, and the output signal are, in particular, electrical signals. Advantageously, the corresponding processing is carried out using the appropriate signal processing unit, suitablely a digital signal processor. Advantageously, the output signal is then output, for example, via the appropriate headphones, so that, in particular, output sound is generated, which is suitablely introduced into the user's ear canal.

[0018] The first and second gain factors are always positive or negative, or can be both, depending on specific requirements. The second gain factor is chosen based on the difference between the level of the first processed signal and the level of the second processed signal. For this purpose, the levels of both processed signals are conveniently determined.

[0019] Due to this process, the sound originating from the user's own speech is modified according to the first amplification factor and perceived accordingly. The sound originating from the speech of other people is perceived in a similarly adjusted manner. If the user is conversing with another person, they may have difficulty understanding the other person, for example, due to a faulty signal processing unit, impaired hearing, and / or unfavorable background noise. In this case, the user will unconsciously speak louder. Consequently, the difference between the level of the first processed signal and the level of the second processed signal changes. As a result, the second amplification factor is adjusted, so that the level of the second processed signal is subsequently increased.Consequently, even if the other person doesn't speak louder, they are easier for the user to understand, thus increasing user comfort and improving conversation. This makes it possible to utilize the Lombard effect, which describes how people in a comparatively noisy environment also (unconsciously) speak louder, even if the person they are speaking to is not susceptible to this effect.

[0020] Preferably, the second amplification factor is configured such that the signal-to-noise ratio (SNR) of the two processed signals to each other, or at least of the second processed signal, exhibits a specific ratio or is at least within a specific range. This further improves speech intelligibility.

[0021] The first amplification factor is preferably predetermined based on any hearing loss of the user. Alternatively, or in combination with this, the first amplification factor is specified by the user, or in particular, adapted to them. Preferably, the first amplification factor is selected based on the ambient noise and / or a classification of the environment.

[0022] The second gain factor is preferably selected such that the level of the first processed signal differs from the level of the second processed signal by less than a threshold value. For example, the second gain factor is determined only at specific times, such as at the beginning of the process and / or when a specific operating mode is set, or when the second signal component is first present. Alternatively, the second gain factor is preferably adjusted continuously, at least as long as the second signal component can be extracted from the input signal.

[0023] The threshold value can be constant or dependent on the user's current situation. For example, the second gain factor is chosen such that the levels of the two signals are equal. Alternatively, the second gain factor is chosen such that the two levels differ only by the threshold value. Preferably, at least the level of the second processed signal is lower than the level of the first processed signal if the level of the second signal component is lower than the level of the first signal component, and vice versa. This prevents an excessive shift in the relative levels.

[0024] For example, the first gain factor is used as the second gain factor, to which a specific value is added depending on the difference. For this, the first processed signal is first created and its level determined. InDepending on this, the second gain factor is then selected. Alternatively, a compression curve can be modified over time, or a time constant can be added to an adaptive compression system. Alternatively, the second gain factor can be calculated based on the first and second signal components, particularly their relative levels, and the knowledge of the first gain factor, so that the difference in the resulting processed signals is less than the threshold. Specifically, the second gain factor is only changed if the difference is greater than the threshold. If, however, the difference is less than the threshold with an initial second gain factor that is, for example, equal to the first, it is more practical to leave the second gain factor unchanged.

[0025] For example, a second preliminary processed signal is first created using the second signal component and a second preliminary gain factor. The second preliminary gain factor is specifically defined / configured in the same way as the first gain factor and is preferably adapted to the user's hearing loss. The level of the second preliminary processed signal is compared to the level of the first processed signal, and the second gain factor is selected based on this comparison. Using the second gain factor, the second processed signal is then created from the second preliminary processed signal. Thus, the second signal component is first processed with the second preliminary gain factor and then with the second gain factor, resulting in the second processed signal.This simplifies the process of adjusting the level of the second processed signal to match the level of the first processed signal. Specifically, the additional adjustment is only performed if the level of the first processed signal differs from the level of the second preliminary processed signal by more than the threshold value. If there is no difference, a second gain factor of "1" or the identity is appropriately used, so that the second preliminary processed signal matches the second processed signal. Conversely, if the difference exceeds the threshold value, the second gain factor is appropriately selected, shifting the second processed signal towards the level of the first processed signal relative to the second preliminary processed signal.Suitablely, the difference between the levels of the two processed signals is subsequently equal to the limit value, so that no excessive shift occurs and thus the ratio of the processed signals to the two signal components is not excessively changed.

[0026] Alternatively or in combination with this, automatic gain control (AGC) is used. This involves using several gain curves to map the input signal to the output signal. The respective gain curve is selected based on the current environment or situation, ensuring that the output signal level remains within predefined limits. Consequently, all sounds of interest to the user are perceptible without excessive amplification. Preferably, the gain curves are at least partially linear and / or continuous, so that the relative levels of sounds contained in the input signal are preserved in the output signal, thus improving intelligibility for the user.

[0027] It is advantageous to use a first gain curve for the first signal component and a second gain curve for the second signal component. In other words, different gain curves, and therefore gain factors, are assigned to the different signal components. Thus, at least in certain sections, the two signal components are amplified differently, so that the difference between the levels of the two processed signals meets a specific requirement defined by the two gain curves. Therefore, the second gain factor, which is determined at least partially by the second gain curve, is chosen as a function of the difference between the levels of the processed signals.Thanks to the automatic gain control, user comfort is improved in any given situation, regardless of whether the environment is relatively loud or quiet. This system utilizes the Lombard effect due to the different gain curves, so that if the user feels they are having difficulty understanding the other person and therefore speaks louder, the louder speech of the other person is amplified, thus improving intelligibility.

[0028] In particular, a background noise level is first determined. This corresponds specifically to the portion of the input signal that is not assigned to either the first or the second signal component. The level of the background noise is then determined. Thus, the level of the background noise in the current situation is assigned to it. The two gain curves are, for example, designed such that they differ at the level of the background noise. Preferably, however, the two gain curves are identical at the level assigned to the background noise in the current situation. Therefore, if neither the user's nor the other person's language is present, it is irrelevant whether this portion of the input signal is assigned to the first or the second signal component. The processing is always identical and thus always results in the same portion in the output signal.Therefore, switching between the two gain curves is possible during pauses in conversation without affecting the output signal. In other words, switching between the gain curves is silent, without any crackling or similar noise. Alternatively, or in combination with this, a gradual adjustment between the gain curves (a "fading" effect) can occur during the switch, preventing the formation of artifacts and further improving user comfort.

[0029] Alternatively, or preferably in combination with this, a user-specific maximum level is predefined, corresponding, for example, to the user's pain threshold or at least a discomfort threshold. This user-specific maximum level is defined, for instance, by the user, an audiologist, or the hearing aid manufacturer. For example, the user-specific maximum level may differ between all users or be the same for some, many, or all users. A maximum level of the first signal component, determined for the current situation, results in the user-specific maximum level. In other words, the first gain curve adjusts the portion of the first signal component that exhibits the maximum level for the current situation so that the corresponding portion of the first processed signal reaches the user-specific maximum level.The maximum level for the current situation is determined continuously and corresponds, in particular, to the maximum of the first signal component in the current situation up to the current time, or, for example, to the average value over a specific period. Advantageously, the first gain curve between the user-specific maximum level and the level associated with the background noise of the current situation is linear.

[0030] Alternatively or in combination, a maximum level of the second signal component determined for the current situation is used to achieve the user-specific maximum level via the second gain curve. In other words, those parts of the second signal component that exhibit the maximum level determined for the current situation will have the user-specific maximum level after amplification. Thus, the maximum levels are assigned to the two signal components by means of the gain curves, although the maximum levels determined for the current situation may differ between the two signal components. Preferably, the two gain curves between the level assigned to the background noise of the current situation and the user-specific maximum level are essentially linear, with the two gain curves having, in particular, different slopes.This facilitates processing, and the user can understand the other person's speech relatively well. Furthermore, the second gain factor is selected based on the difference between the level of the first processed signal and the level of the second processed signal, by appropriately choosing the gain curves. However, an implicit determination of the difference is not necessary.

[0031] For example, the second signal component corresponds only to the speech of a single person. If, for instance, several people are present, a separate second amplification factor is selected for each. However, it is particularly advantageous for the second signal component to correspond to the speech of multiple people, ideally to the speech of all people present in the current situation. Thus, those parts of the input signal that result from the speech of other people are assigned to the second signal component. This simplifies processing.

[0032] 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.

[0033] The hearing aid includes a microphone. This microphone is, for example, omnidirectional, or its directional characteristics can be adjusted. For this purpose, the microphone preferably has two or more microphone units. The microphone is designed and configured to detect ambient sound. Advantageously, an input signal is generated by the microphone when ambient sound is detected. The hearing aid also includes a signal processing unit, which is preferably connected to the microphone. In particular, the input signal is fed to the signal processing unit during operation.

[0034] The hearing aid operates according to a method in which the input signal is generated from ambient sound. A first signal component and a second signal component are extracted from the input signal, the first signal component corresponding to the speech of one user and the second signal component corresponding to the speech of another person. A first processed signal is generated from the first signal component and a first gain factor, and a second processed signal is generated from the second signal component and a second gain factor. The two processed signals are combined to form an output signal. The first gain factor is selected based on the difference between the level of the first signal component and the level of the second signal component. Advantageously, the signal processing unit is suitable, and in particular designed and configured, to carry out at least part of the method.

[0035] The further training and advantages explained in connection with the procedure can also be applied analogously to the hearing aid and to each other, and vice versa.

[0036] Exemplary embodiments of the invention are explained in more detail below with reference to a drawing. The drawing shows: Fig. 1 schematically simplified a hearing aid, Fig. 2 a method for operating the hearing aid, Fig. 3 schematically a type of processing of a first signal component and a second signal component during the method, and Fig. 4, 5 amplification curves used in an alternative type of processing.

[0037] Corresponding parts are marked with the same reference symbols in all figures.

[0038] In Figure 1The schematically simplified representation comprises a hearing aid 2. The hearing aid 2 has a housing 4, inside which a microphone 6 is arranged. 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, which includes a control unit 10, is connected downstream of the microphone 6. A receiver 12 is connected downstream of the signal processing unit 8, by means of which, when used as intended, sound can be emitted into the ear canal of the user (not shown in detail).

[0039] In Figure 2A method 14 for operating the hearing aid 2 is shown, which is carried out at least partially by means of the signal processing unit 8. In a first step 16, an input signal 20 is generated from an ambient sound 18. For this purpose, the ambient sound 18 arriving at the microphone 6 from outside the housing 4 is detected by means of the microphone 6 and converted into the electrical input signal 20, which is then sent to the signal processing unit 8. The ambient sound 18 consists of three components, one of which represents the speech 22 of the user. Another component of the ambient sound 18 is present due to a conversation partner and is thus the speech 24 of another person. The third component is caused by other noise sources 26.

[0040] In a subsequent second step 32, a first signal component 32 and a second signal component 34 are extracted from the input signal 20 using a signal processing unit 30 of the signal processing unit 10. The remainder of the input signal 20 is assigned to a third signal component 36. The first signal component 32 corresponds to the user's speech 22, and the second signal component 34 corresponds to the speech 24 of the other person. If several people are speaking, their speech is also assigned to the second signal component 34. In other words, the second signal component 34 corresponds to the speech 24 of multiple people. For the signal division, a spatial analysis is used, for example, to determine the origin of the individual components of the ambient sound 18. For this purpose, the directional characteristic of the microphone 6 is set / checked.

[0041] In a subsequent third step 38, the first signal component 32 is processed using a first amplification factor 40, resulting in a first processed signal 42. Specifically, the first signal component 32 is multiplied by the first amplification factor 40. The first amplification factor 40 is predetermined and selected based on the user's hearing loss. Due to the processing with the first amplification factor 40, the level of the first signal component 32 is increased, resulting in a higher level for the first processed signal 42, as schematically shown in Figure 3.

[0042] Furthermore, the second signal component 34 is first multiplied by a second preliminary gain factor 44, resulting in a second preliminary processed signal 46. The second preliminary gain factor 44 is equal to the first gain factor 40, thus initially providing amplification based on the user's hearing loss. Due to the processing, the level of the second preliminary processed signal 46 is also increased, as shown in Figure 3. Subsequently, the level of the first processed signal 42 differs 48 from the level of the second preliminary processed signal 46. In the example shown, the difference 48 is greater than a limit value 50, and the level of the second preliminary processed signal 46 is lower than the level of the first processed signal 42.

[0043] The second preliminary processed signal 46 is then multiplied by a second gain factor 52 to create a second processed signal 54. The second gain factor 52 is chosen such that the level of the second processed signal 54 differs from the level of the first processed signal 42 by exactly the threshold 50, where the level of the second processed signal 54 is lower than the level of the first processed signal 42. If the level of the second preliminary processed signal 46 differs from the level of the first processed signal 42 by less than the threshold 50, the second gain factor 52 is set to "1". Thus, the second preliminary processed signal 46 corresponds to the second processed signal 54.

[0044] In summary, the first gain factor 40 is specified, and the second gain factor 52 is chosen such that the level of the first processed signal 42 differs from the level of the second processed signal 54 by less than the limit value 50. The second gain factor 52 is also chosen depending on the difference 48 between the level of the first processed signal 42 and the level of the second processed signal 54, specifically such that the difference 48 is less than the limit value 50. For this purpose, the second preliminary processed signal 46 is first created using the second signal component 34 and the second preliminary gain factor 44, and its level is then compared with the level of the first processed signal 42. InDepending on the comparison, the second gain factor 52 is then selected. If the level of the second preliminary processed signal 46 is greater than the level of the first processed signal 42 and the difference 48 is greater than the limit 50, the second gain factor 52 is selected such that the levels of the two processed signals 42, 54 differ by less than the limit 50, but the level of the second processed signal 54 is greater than the level of the first processed signal 42.

[0045] The third signal component 36 is processed with a third amplification factor 56, resulting in a third processed signal 58. The third amplification factor 56 is predetermined depending on the user's hearing loss and the current situation.

[0046] In an alternative configuration, instead of direct multiplication by the gain factors 40, 44, 52, an automatic gain control is used. Here, the first signal component 32 is assigned a value in Figure 4 The first amplification curve 62 and the second signal component 34 are shown in Figure 5 The second gain curve 64 shown is assigned to the first. The first gain factor 40 and the second gain factor 52 are at least implicitly specified by means of the two gain curves 62 and 64. The first gain curve 62 indicates which level should be used for the first processed signal 42 for a given level of the first signal component 32. Likewise, the second gain curve 64 indicates to which level the respective level of the second signal component 34 should be mapped in order to maintain the second processed signal 54.

[0047] The two gain curves 62 and 64 are designed and adapted to the current situation such that they are identical at the level of a background noise 66. This background noise 66 arises from the ambient noise of the user's current situation. In the example shown, the unit 40 of the level of both the first signal component 32 and the second signal component 34 is thus assigned the unit 60 as the level of the first processed signal 42 and the second processed signal 54, respectively. Below the level of the background noise 66, the two gain curves 62 and 64 are identical and linear. At this level, the two gain curves 62 and 64 have a slope of 1 and are shifted parallel to the identity 68 (identical figure), which is shown as a dotted line.In summary, the two gain curves 62 and 64 are chosen such that they are identical at the level associated with the background noise 66 of the current situation. Therefore, if neither the user nor other persons are speaking, or if the level of the respective speech 22 or 24 is lower than the level of the background noise 66, it is irrelevant which of the two gain curves 62 or 64 is selected, and the first processed signal 42 then corresponds to the second processed signal 54.

[0048] Furthermore, a maximum level of 70 for the first signal component 32 is determined for the current situation, which in the example shown almost reaches the value 70. To determine the maximum level 70 of the first signal component 32, the current level of the first signal component 32 is recorded for a specific period, such as 10 seconds, and the maximum of this is used as the maximum level 70 of the first signal component 32 for the current situation. A user-specific maximum level 72 is assigned to this maximum level 70 of the first signal component 32, which in the example shown has a value of 80. The user-specific maximum level 72 is either specified by the manufacturer of the hearing aid 2 or adjusted to the user, for example, by an audiologist. The user-specific maximum level 72 represents the discomfort threshold for the user.Thus, those portions of the first signal component 32 that exhibit the maximum level 70 of the first signal component 32 for the current signal are assigned the user-specific maximum level 20. Therefore, two points are defined for the first gain curve 62: the level of the background noise 66, which is assigned the value 60, and the maximum level 70 of the first signal component 32, which is assigned the user-specific maximum level 72. Between these two points, the curve of the first gain curve 62 is linear. At higher levels, the curve of the first gain curve 62 is also linear, but the slope is reduced.

[0049] A maximum level 74 is also determined for the second signal component 34 for the current situation. This is done in the same way as the determination of the maximum level 70 of the first signal component 32. In the example shown, the maximum level 74 of the second signal component 34 has the value 50. The user-specific maximum level 72, i.e., the value 80, is also assigned to this. The second gain curve 64 exhibits a linear progression between the two points defined by the background noise 66 and the maximum level 74 of the second signal component 34. However, since the maximum level 70 of the first signal component 32 is greater than the maximum level 74 of the second signal component 34, the slope of the second gain curve 64 is increased.Above the maximum level 74 of the second signal component 34, the course of the second gain curves 64 is again linear, namely up to the values ​​specified by means of the maxima of 90, as is also the case with the first gain curve 62.

[0050] In summary, the two gain curves 62, 64 will be chosen such that the maximum level (70) of the first signal component 32 determined for the current situation leads to the user-specific maximum level 72.

[0051] The maximum level 74 determined for the current situation of the second signal component 34 also leads to the user-specific maximum level 74.

[0052] In the third work step 38, regardless of the specific design, the first processed signal 42 is created using the first signal component 32 and the first gain factor 40, and the second processed signal 54 is created using the second signal component 34 and the second gain factor 52.

[0053] In a subsequent fourth step 76, the third processed signal 58, the second processed signal 54, and the first processed signal 42 are added together by an adder 78 of the signal processing unit 10 to form an output signal 80, which is output by the adder 65. Thus, the first and second processed signals 42 and 54 are combined to form the output signal 80.

[0054] In a subsequent fifth step 82, the output signal 80 is presented to the user via the receiver 12. If the user has difficulty understanding the speech 24 of the other person, they unintentionally speak louder, as described by the Lombard effect. This leads to a stronger amplification of the second signal component 34, so that the second processed signal 54 has a higher level. Depending on the configuration, this occurs even with a slight increase in the volume of the other person's speech 24, or even when the other person is not speaking louder. Thus, intelligibility for the user is improved.

[0055] The invention is not limited to the embodiments described above. Rather, other variants of the invention can also be derived by a person skilled in the art without departing from the subject matter of the invention. In particular, all individual features described in connection with the individual embodiments can also be combined with one another in other ways without departing from the subject matter of the invention. Reference symbol list

[0056] 2 Hearing aid 4 Housing 6 Microphone 8 Signal processing unit 10 Control unit 12 Receiver 14 Procedure 16 First processing step 18 Ambient sound 20 Input signal 22 User's language 24 Another person's language 26 Other noise source 28 Second processing step 30 Splitting unit 32 First signal component 34 Second signal component 36 Third signal component 38 Third processing step 40 First gain factor 42 First processed signal 44 Second preliminary gain factor 46 Second preliminary processed signal 48 Difference 50 Limit value 52 Second gain factor 54 Second processed signal 56 Third gain factor 58 Third processed signal 60 Automatic gain control 62 First gain curve 64 Second gain curve 66 Background noise 68 Identity 70 Maximum Level of the first signal component 72 User-specific maximum level 74 Maximum level of the second signal component 76 Fourth step 78 Adder 80 Output signal 82 Fifth step

Claims

1. Method (14) for operating a hearing aid (2) in which: - an input signal (20) is generated from an ambient sound (18); - a first signal component (32) and a second signal component (43) are extracted from the input signal (20), wherein the first signal component (32) corresponds to speech (22) of a user and the second signal component (34) corresponds to speech (24) of another person; - a first processed signal (42) is generated from the first signal component (32) and a first gain factor (40); - a second processed signal (54) is generated from the second signal component (34) and a second gain factor (52); and - the two processed signals (42, 54) are combined to form an output signal (80), wherein the second gain factor (52) depends on a difference (48) between the level of the first processed signal (42) and the level of the second processed signal. (54) is elected.

2. Method (14) according to claim 1, characterized by that the first gain factor (40) is specified, and the second gain factor (52) is chosen such that the level of the first processed signal (42) differs from the level of the second processed signal (54) by less than a limit value (50).

3. Method (14) according to claim 1 or 2, characterized by that a second preliminary processed signal (46) is created using the second signal component (34) and a second preliminary gain factor (44), the level of which is compared with the level of the first processed signal (42), the second gain factor (52) is chosen depending on the comparison, and the second processed signal (54) is created using the second preliminary processed signal (46) and the second gain factor (52).

4. Method (14) according to any one of claims 1 to 3, characterized by thatAn automatic gain control is used, whereby a first gain curve (62) is assigned to the first signal component (32) and a second gain curve (64) is assigned to the second signal component (34).

5. Method (14) according to claim 4, characterized by that the two gain curves (62, 64) are the same at the level assigned to a background noise (66) of the current situation.

6. Method (14) according to claim 4 or 5, characterized by that the two gain curves (62, 64) are chosen such that a maximum level (70) of the first signal component (32) determined for the current situation leads to a user-specific maximum level (72), and that a maximum level (74) of the second signal component (34) determined for the current situation leads to the user-specific maximum level (72).

7. Method (14) according to any one of claims 1 to 6, characterized by thatthe second signal component (34) corresponds to speech (24) from several people.

8. Hearing aid (2) comprising a microphone (6) for detecting ambient sound (18) and a signal processing unit (8), and operated according to a method (14) according to any one of claims 1 to 7.

Citation Information

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