METHOD FOR OPERATING A HEARING SYSTEM AND HEARING SYSTEM

Description

[0001] The invention relates to a method for operating a hearing system and a corresponding hearing system.

[0002] In its most general form, a hearing aid consists of a signal processor and a receiver, which is also referred to as a loudspeaker. An input signal is fed into the signal processor, which then modifies it to generate an output signal, which is essentially a modified version of the input signal. For example, the input signal might be amplified by a specific gain factor. Hereafter, both the input and output signals will be referred to as signals. The output signal is then reproduced by the receiver. Both the input and output signals are electrical signals. The receiver then converts the output signal into an audible sound signal.

[0003] Optionally, the hearing system also features a microphone that generates the input signal by picking up one or more sound signals from the environment and converting them into the input signal. Such a hearing system with an additional microphone is also called a hearing aid and is used, for example, to provide hearing assistance to a person with hearing loss by amplifying sound signals from the environment and delivering them to the person as amplified sound signals.

[0004] In hearing aids, it is possible to use non-linear amplification instead of simple linear amplification. This is achieved, for example, with a compressor, which provides and applies a compression scheme that adjusts the amplification factor appropriately when modifying the input signal, i.e., when generating the output signal. The compressor's behavior, i.e., the compression scheme, is defined by one or more parameters, more precisely, compression parameters.

[0005] US Patent 7,773,763 B2 describes a hearing aid in which different programs are associated with different environment types. Each program contains specific values ​​for certain hearing aid parameters to achieve optimal signal quality in a given environment. The hearing aid can classify the current environment, assign it to an environment type, and then select a suitable program with the corresponding optimal parameters.

[0006] In DE 10 2005 061 000 A1 a method is described in which a switch is made between two compression algorithms depending on a classification result.

[0007] In DE 10 2010 041 740 A1 a method is described in which different dynamic compressions are performed on different frequency bands.

[0008] US Patent 6,198,830 B1 describes a method for amplifying input signals of a hearing aid and a circuit for carrying out the method.

[0009] US patent 2016 / 381468 A1 describes a method for compressing the dynamic range in an audio signal.

[0010] EP 2 908 550 A1 describes a hearing aid device with a sensor element.

[0011] Against this background, an object of the invention is to provide an improved method for operating a hearing system and a corresponding hearing system. In particular, the dynamic range of a hearing system with automatic gain control is to be improved.

[0012] The problem is solved according to the invention by a respective method with the features according to claims 1 and 2, and by a hearing system with the features according to claim 14. Advantageous embodiments, further developments, and variants are the subject of the dependent claims. The descriptions relating to the method also apply mutatis mutandis to the hearing system and vice versa.

[0013] The procedure serves to operate a hearing aid and is therefore an operating procedure. During operation, the hearing aid is worn by a user, specifically on, in, or behind the ear, i.e., generally on the head and in the area of ​​one or both ears. At any given time, the hearing aid and the user are in a current acoustic environment. This means that the environment around the user and the hearing aid forms a specific acoustic backdrop at that particular moment.

[0014] The hearing system features signal processing that generates an output signal from an input signal by amplifying the input signal with a gain factor. The signal processing thus acts as an amplifier. The signal processing is preferably integrated into the hearing system. Preferably, the hearing system is a hearing aid, and particularly preferably a hearing aid for a hearing-impaired person. As a hearing aid, the hearing system has at least one microphone that picks up acoustic signals, i.e., sound signals from the environment, and converts them into an input signal. The hearing system also has a receiver through which the output signal, i.e., the amplified input signal, is emitted. The receiver is also referred to as a loudspeaker.

[0015] The amplification factor is set by an automatic gain control, which is a key component of the signal processing. Automatic gain control is also known as AGC (automatic gain control). "Automatic" means that no user input is required to adjust the gain; this happens automatically during operation. In addition to automatic gain control, a suitable hearing aid may also feature a manual gain control, allowing the user to further adjust the amplification factor as desired.

[0016] The automatic gain control system incorporates a dynamics processor that can be operated with a dynamics scheme. This scheme defines the gain factor as a function of the input signal level. The dynamics scheme thus specifies which gain factor is used for a given input level. The dynamics scheme is also referred to as a dynamics algorithm. Examples of dynamics schemes include compression or expansion schemes in general, and specifically AGCi (automatic gain control input-dependent) and ADRO (adaptive dynamic range optimization). Several dynamics schemes are suitable for this application, with AGCi being particularly preferred.

[0017] A key characteristic of a dynamics processor is that the gain factor is not constant; rather, the processor results in non-linear amplification. Specifically, the dynamics processor regulates the amplification through signal processing based on a control variable. This control variable could be, for example, the input level or, alternatively, the output level. Both options have specific advantages and disadvantages. The respective level is measured, for example, by a level detector, which is ideally integrated into the hearing system.

[0018] Generally, both the input and output signals exhibit a dynamic range, meaning a level that lies within a specific range, from a minimum to a maximum level. The level of the input signal is also referred to as the input level, and the level of the output signal as the output level. Similarly, the dynamic range of the input signal is called input dynamic range, and the dynamic range of the output signal is called output dynamic range. The dynamic range of the output signal depends on the dynamic range of the input signal as well as on any modifications made by signal processing. In certain applications, it is desirable to limit (i.e., compress) or increase (i.e., expand) the dynamic range of the output signal. In a hearing aid for a hearing-impaired person, the dynamic range is appropriately adjusted to the reduced hearing ability of the hearing-impaired person compared to someone with normal hearing.The reduced hearing ability is limited to a specific level range, which then conveniently defines the dynamics of the output signal. The dynamics of the input signal are then mapped to this level range by generating the output signal with a suitably appropriate dynamic range, in order to achieve maximum sound quality. Depending on the design, it may then be advantageous to apply compression or expansion, or even a combination of both.

[0019] During operation, the hearing aid runs in a specific program, which is assigned a dynamic range. The program thus pre-configures the dynamic processor in a predefined way. The program and the dynamic range are appropriately tailored to the current acoustic environment to ensure optimal output, i.e., output quality. Therefore, the dynamic processor operates in this one program with its associated dynamic range during operation.

[0020] A key concept of the invention is that at least one situation parameter is determined which characterizes, i.e., in particular quantifies or qualifies, the current acoustic environment. During operation, the associated dynamic scheme is modified in one program depending on the situation parameter, so that the dynamic processor operates with a modified dynamic scheme. The dynamic scheme specified by the program for the current acoustic environment is thus optimized situationally. In other words, the dynamic scheme for automatic gain control is modified depending on the situation. This is based on the consideration that the acoustic environment can change, and as a result, the selected dynamic scheme is no longer optimal, although the change may not be sufficiently significant or long-lasting to necessitate a program change, e.g.,to justify the change from a speech to a music program. An abrupt change in the dynamic range is thus advantageously avoided; instead, the dynamic range actually assigned to the environment is adapted to the situation and thus further developed or refined, so to speak, as part of a fine-tuning process. This improves the output of the hearing system in a way that is pleasant for the user.

[0021] In this case, the dynamics are not adjusted solely depending on the input level or the output level; rather, the situation parameter is preferably additionally supplied to the automatic amplification unit and used as an additional control or regulation variable.

[0022] The situation parameter is preferably determined by means of a detector. The detector is preferably part of the hearing system. Alternatively, the situation parameter is determined externally with respect to the hearing system and transmitted to the hearing system, e.g. via a data connection.

[0023] A particularly preferred configuration is one in which the dynamics processor is a compressor, resulting in a compression dynamics scheme. Another advantageous configuration is one in which the dynamics processor is an expander, resulting in an expansion dynamics scheme. Compressors and expanders typically operate in opposite ways. While a compressor usually limits the gain above a certain threshold, an expander typically increases the gain above a certain threshold. In other words, a compressor amplifies low-level signals more than high-level signals, and conversely, an expander amplifies high-level signals more than low-level signals. Overall, a compressor reduces the dynamic range of a signal, and a compressor increases the dynamic range of a signal.

[0024] An advantageous configuration combines a compressor and an expander, either as two separate dynamic processors or together as a single dynamic processor. In a first configuration, the compressor and the expander are used independently, so that only one of them is active at any given time, for example, depending on the situational parameters or a selected program of the hearing system. In a second configuration, the compressor and the expander are active simultaneously in a so-called hybrid mode, preferably such that the compressor reduces the overall level and the expander increases the dynamic range again at an operating point, i.e., within a limited frequency and / or level range. In other words, the expander is used to locally counteract what would otherwise be a global compression.A reverse configuration, in which the expander acts globally and the compressor only locally, is also fundamentally possible and suitable.

[0025] In any given situation, it is technically possible to use either a compressor, an expander, or a combination of both. Which of these three approaches is audiologically most appropriate depends on the specific situation. While a combination of compressor and expander may initially seem contradictory, it is technically feasible and, in some situations, even beneficial.

[0026] The following assumes, without limitation of generality, that either a compressor or an expander is used. However, the statements also apply analogously to configurations in which both a compressor and an expander are used, and in which, depending on the specific design, several dynamic schemes may result, to which the described preferred configurations are then expediently applied.

[0027] Modification within a given program is ultimately a specific adaptation of the program itself to individual deviations of the current acoustic environment from an idealized or standard environment predicted during the program's creation. A particular challenge in operating a hearing aid is that different situations can occur simultaneously in different environments, requiring mutually exclusive programs. A particularly important example is the presence of speech in the environment. To make this speech as intelligible as possible for the user, a program that enhances speech intelligibility is appropriately selected. In this case, the realistic reproduction of other sound signals or noises is of secondary importance; the primary goal is to make speech recognizable to the user.For this purpose, a special dynamic range optimized for maximum speech intelligibility is selected and used. Conversely, in a speech-free environment, the aim is to reproduce the acoustic environment as realistically as possible, thus achieving the best possible sound quality. The best possible sound quality means, in particular, that any hearing impairment of the user is compensated for as effectively as possible, i.e., maximum hearing loss compensation is achieved. This is especially important with music, which can be significantly distorted by a dynamic range designed for improved speech intelligibility. The same applies to other sound signals in the environment, which can sometimes be distorted so severely that they are no longer recognizable or identifiable to the user.This leads to particular problems when both speech and other sound signals are present in the environment, i.e., when the acoustic environment generally comprises several different sound signals, each requiring different programs for optimal output—for example, a speech program on the one hand and a music program on the other. In principle, a compromise is possible by using a program that is not optimized in just one direction, but rather attempts to reproduce all sound signals at least partially appropriately. However, it becomes clear that none of the sound sources will then be output optimally, meaning that speech, in particular, will never be optimally intelligible with such a program.

[0028] This problem is circumvented here by fine-tuning the dynamic range within the program. The dynamic range is modified depending on the situation by determining at least one situation parameter, and preferably several situation parameters, particularly on a recurring basis, to ensure optimal dynamic range processor settings at any given time. Thus, the dynamic range processor is adapted to the specific situation. In a suitable configuration, the situation parameter is determined by the hearing system itself.

[0029] In particular, this can advantageously resolve the conflict of objectives described above between speech intelligibility and sound quality. In a particularly advantageous configuration, the situation parameter indicates whether speech is present in the environment or not, and the dynamic range is modified to prioritize speech intelligibility if speech is present, and otherwise to prioritize sound quality. Thus, a program optimized for maximum sound quality, preferably a music program, is selected from the outset. In the case of speech, speech intelligibility is then prioritized, and the program, specifically the dynamic range, is modified to achieve maximum speech intelligibility, particularly without changing the program itself. This modification is expediently maintained only as long as speech is present.The modification occurs automatically, advantageously requiring no user input. The user also doesn't need to select a specific speech, sound, or music program. The system reacts directly to the current acoustic environment, eliminating the need for program selection. Therefore, the program is not static but dynamic, adapting to the situational parameters through the modifications performed. This ensures that the user generally experiences maximum sound quality, and in situations involving speech, maximum speech intelligibility. To determine whether speech is present, a speech detector is appropriately used, specifically examining the input signal for the presence of speech.

[0030] The dynamic scheme defines the behavior of the dynamic processor during operation and is defined by one or more parameters, specifically dynamic parameters, such as compression parameters and / or expansion parameters. Suitable dynamic parameters include, in particular, a compression or expansion threshold, also known as the knee point; a compression or expansion ratio, often referred to simply as compression or expansion; an attack time; and a release time. The attack and release times of the dynamic processor are also referred to as time constants. The dynamic scheme is modified by changing the attack time and / or the release time.

[0031] Preferably, the dynamic scheme is modified by changing at least one further parameter of the dynamic scheme, wherein the parameter is selected from a set of parameters comprising: a knee point, a compression ratio in the case of a compressor, and an expansion ratio in the case of an expander. According to the invention, the turn-on time or the turn-off time, or both, are changed depending on the situation. However, the other parameters are also suitable for modifying the dynamic scheme.

[0032] Furthermore, the program in question does not switch between two different dynamic schemes, but rather uses the same dynamic scheme, i.e., the same dynamic algorithm, whose parameters are adapted depending on the situation. This ensures that the underlying concept of the dynamic scheme is preserved and only its specific implementation is modified. This contrasts with the aforementioned DE 10 2005 061 000 A1, which switches between fundamentally different dynamic schemes depending on the situation. In contrast, the present program dynamically modifies the parameterization of the dynamic scheme.

[0033] Dynamic range schemes for maximizing sound quality and speech intelligibility are generally known. For maximum sound quality, for example, a linear amplification result is sought, so that the dynamic range scheme only applies compression at high input levels, whereas no compression occurs at low and medium input levels, with a compression ratio of approximately 1. A knee point, which marks the transition to a higher compression ratio, is then positioned relatively high, i.e., compared to a knee point of a dynamic range scheme for maximum speech intelligibility. Here, for example, the level range of the input signal containing speech is mapped as closely as possible to the level range audible to the user.

[0034] The situation parameter is the signal-to-noise ratio of the current acoustic environment, and especially of the input signal. The dynamic range is modified based on this ratio. In other words, the dynamic range is adjusted according to the relative intensity of background noise compared to the desired signal—the sound signal the user wants to perceive. This is based on the assumption that with a high signal-to-noise ratio, the desired signal is already sufficiently intelligible for the user, and therefore only minor adjustments, particularly compression, are necessary to improve sound quality.Conversely, a low signal-to-noise ratio indicates a highly noisy environment, so the dynamic range is specifically modified to amplify the desired signal relative to the interference, thereby making it more intelligible for the user. A noise detector is conveniently used to determine the signal-to-noise ratio. The input signal is fed into the noise detector, which then uses the signal-to-noise ratio of the environment to determine the signal-to-noise ratio.

[0035] In the aforementioned embodiment of the invention, the signal-to-noise ratio is the ratio of a signal originating from a speaker to noise originating from one or more other sound sources in the environment. These other sound sources include, for example, other people, such as an audience, or devices, such as a vehicle in which the user and the speaker are located. The dynamic scheme comprises two time constants: a turn-on time and a turn-off time. In the case of a negative signal-to-noise ratio, at least one of the time constants is reduced, and conversely, in the case of a positive signal-to-noise ratio, at least one of the time constants is increased. "A time constant is reduced" means that a faster time constant is set. Similarly, "A time constant is increased" means that a slower time constant is set.Generally, with regard to the switch-on time, "fast" is understood to mean a time of up to 10 ms and "slow" a time of 100 ms to 1 s. With regard to the switch-off time, "fast" is generally understood to mean a time of up to 100 ms and "slow" a time of 500 ms to 1 s.

[0036] Preferably, in the case of a negative signal-to-noise ratio, both time constants are reduced, and conversely, in the case of a positive signal-to-noise ratio, both time constants are increased. This is based on the consideration that environments with a negative signal-to-noise ratio are very tiring for the user. In such a situation, the user particularly benefits from gaps in the input signal, i.e., from short periods in which the noise is very low, where "short" is understood to mean a duration of at most one second. These gaps are expediently amplified by modifying the dynamic range depending on the situation, using correspondingly fast time constants. "Fast" in this context refers specifically to a time constant of at most 100 ms.This approach improves speech intelligibility in high noise environments, but leads to poorer sound quality. Therefore, this approach is avoided when noise levels are low, and instead comparatively long time constants are chosen, especially time constants longer than 100ms, so that the adjustment, especially the compression, is weaker overall.

[0037] In a further advantageous embodiment, an additional situation parameter is an environment class, such that the dynamic range is modified depending on the environment class. The environment class is selected, in particular, from a set of classes including: speech, music, and background noise. The situation parameter is determined, in particular, by means of a detector configured as a classifier. The classifier is expediently part of the signal processing and determines the environment class, especially based on the input signal. If the environment class is music, the dynamic range is modified more conservatively; that is, for example, less compression is applied than in the case of speech. For speech, the dynamic range is then modified to maximize speech intelligibility. This is advantageously applicable in concerts or similar environments where speech is not present.The dynamic scheme is then modified with a view to achieving the most realistic reproduction of music, rhythms and melodies possible; in particular, the aim is to achieve the most linear amplification possible for the user.

[0038] In a further advantageous embodiment, an additional situation parameter is a movement pattern, such that the dynamic scheme is modified depending on movement of the hearing system. The movement pattern is selected from a set of movement patterns, including: a resting position, a walking motion, a running motion, and a journey. The movement pattern is determined, in particular, by means of a detector configured as a motion sensor, preferably an accelerometer. A distinction is made between movement and rest. The movement is expediently further subdivided into different types of movement, e.g., according to speed. For example, it is recognized whether the user is standing still, walking, or driving in a car.This is based on the idea that speech intelligibility is generally of varying importance in different environmental situations, and therefore the dynamic scheme is appropriately modified depending on the user's state of movement or behavior.

[0039] In a further advantageous embodiment, an additional situation parameter is the stationarity of the current acoustic environment and, in particular, of the input signal, such that the dynamic scheme is modified depending on the stationarity of the current acoustic environment. To determine the stationarity, a stationarity detector is expediently used, to which, in particular, the input signal is fed, and the stationarity is then determined based on this signal. The stationarity is, in particular, a measure of the dynamics of one or more sound sources in the environment, i.e., a measure of the time-dependent change in the level of these sound sources. The stationarity thus indicates the dynamics of the time-dependent level.

[0040] In a further advantageous embodiment, an additional situation parameter is the diffuseness of the current acoustic environment, such that the dynamic scheme is modified depending on the diffuseness of the current acoustic environment. To determine the diffuseness, a diffuseness detector is expediently used, to which the input signal is fed, and the diffuseness is then determined based on this signal. Diffuseness is similar to the stationarity described above in that it is a measure of the variation in the level in the environment, but now it is not time-dependent, but rather location-dependent. Diffuseness thus indicates how directional the environment is, i.e., whether sound signals approach the user from a specific direction (directed) or uniformly from all directions (diffuse). Diffuseness therefore indicates the dynamics of the direction-dependent level. A spatially very homogeneous level thus leads to high diffuseness.

[0041] In a further advantageous embodiment, an additional situational parameter is the distance of the hearing system, in particular any microphone present, to a sound source, especially a speaker, in the current acoustic environment, such that the dynamic range is modified depending on the distance. The distance is determined, in particular, by means of a rangefinder as a detector. Since the hearing system is worn by the user, the distance between the hearing system and the sound source also corresponds to the distance between the user and the sound source. This is based on the consideration that more distant sound sources are usually less intelligible, and therefore the dynamic range is expediently modified in this case to improve speech intelligibility.

[0042] In a further advantageous embodiment, an additional situation parameter is the reverberation time of the current acoustic environment, such that the dynamic scheme is modified depending on the reverberation time. The dynamic scheme has two time constants: a turn-on time and a turn-off time. The reverberation time is determined repeatedly, and if the reverberation time increases, at least one of the time constants is increased; conversely, if the reverberation time decreases, it is decreased. This is based on the idea that in environments with strong reverberation, rapid adaptation, and especially rapid compression—in shorter time constants—are rather disadvantageous. The reverberation time is expediently determined by means of a detector to which the input signal is fed, allowing the detector to determine the reverberation time by analyzing the input signal.

[0043] A combination of the different concepts described above is particularly advantageous. In a suitable embodiment, several situation parameters are determined and used to control the automatic gain control. In another suitable embodiment, the same situation parameter is determined redundantly using several identical or different detectors.

[0044] In a further advantageous embodiment, the single program is a universal program. The universal program is characterized in particular by the fact that it is not adapted to a specific, predefined situation or used in a particular, limited application area, but rather is used in several different situations. In a suitable embodiment, the hearing system is operated exclusively using this single program, thus eliminating the need for different programs, each specifically adapted and optimized for different environments. Instead, adaptation to a changing acoustic environment occurs solely through the situation-dependent modification of the dynamic range. In another suitable embodiment, the hearing aid, in addition to the universal program, also features one or more specialized programs, each adapted for specific, concrete situations.Preferably, each special program is user-configurable and is preferably not set automatically. While the special programs are unchanging, i.e., they specify fixed and predetermined parameter values ​​for each particular situation, both for the dynamic scheme and generally, the universal program is variable, at least due to its situation-adapted dynamic scheme. The universal program is therefore generally also referred to as the automatic program, since it automatically adapts to the environment, whereas the individual special programs are static. In the universal program, the situation parameter is fed directly to the automatic gain control and used to control the gain factor. Preferably, the possibility of manual gain adjustment is also omitted.Manual adjustment by the user is only possible by selecting a specialized program. All adjustments are performed automatically and situationally based on the universal program, by modifying the dynamic scheme according to the situation parameter.

[0045] In another embodiment of the invention, the dynamic scheme is modified depending on the frequency. This means, in particular, that one or more parameters of the dynamic scheme are modified differently for different frequencies, depending on the situation. The input signal has a frequency spectrum, which is divided into at least two frequency ranges. A frequency range is, in particular, a single frequency or a frequency interval. Advantageously, the input signal is divided into several successive frequency ranges by means of a filter bank in the signal processing. The frequency ranges preferably do not overlap, apart from a potentially minor overlap at the edges due to technical reasons. Depending on the situation parameter, the dynamic scheme is now modified not only for the entire frequency spectrum in the same way, but specifically for the individual frequency ranges.The previously described method, in which the dynamic range is modified, is thus carried out in parallel across multiple frequency ranges. This allows for significantly improved compression or expansion. A particular advantage is that the dynamic range can be specifically modified for certain frequency ranges, ensuring optimal adaptation to the current acoustic environment. Frequency-dependent modification of the dynamic range is suitable for all the described situation parameters. A configuration that performs compression or expansion in different frequency ranges as needed, or a combination of both, is also fundamentally suitable.

[0046] The parameter used to modify the dynamic scheme is therefore changed differently in different frequency ranges depending on the situation in the frequency-dependent modification. According to the alternative embodiment, the dynamic scheme is modified according to the invention by changing the switch-on time or the switch-off time of the dynamic scheme, or both, depending on the frequency. In other words, the switch-on time or the switch-off time, or both parameters, are changed independently of each other for different frequency ranges, so that different time constants may also result for different frequency ranges. This allows specific frequency ranges to be modified with respect to the time constant without having to perform the same modification for all other frequency ranges.

[0047] In the aforementioned alternative embodiment of the invention, a distinction is made between speech frequency ranges and speech-free frequency ranges, i.e., between frequency ranges containing speech and frequency ranges that do not. A suitable speech frequency range is a frequency range from 800 Hz to 4 kHz or a subset thereof. Suitable speech-free frequency ranges are correspondingly frequency ranges below 800 Hz and above 4 kHz. Alternatively, a more detailed division of the frequency spectrum is performed in which the speech frequency ranges correspond to individual sounds or formants, and all remaining frequencies then constitute speech-free frequency ranges. A parameter of the dynamic scheme is changed frequency-dependently by setting the parameter on a respective frequency range depending on whether this frequency range is a speech frequency range or a speech-free frequency range.The frequency-dependent modification thus corresponds to a speech-dependent modification. Overall, this advantageously discriminates between speech frequency ranges and speech-free frequency ranges, and the dynamic range is optimally modified for each type of frequency range. This results, in particular, in improved speech intelligibility. The speech-dependent modification of the dynamic range is also suitable for all the described situation parameters.

[0048] In the aforementioned alternative embodiment of the invention, the speech-dependent modification is combined with the previously described embodiment, in which the situation parameter is a signal-to-noise ratio of the current acoustic environment and, in particular, of the input signal. In the case of a negative signal-to-noise ratio, at least one of the time constants of the dynamic scheme is reduced on a speech frequency range, as already described, and conversely, in the case of a positive signal-to-noise ratio, at least one time constant is increased. On a speech-free frequency range, conversely, in the case of a negative signal-to-noise ratio, at least one time constant is increased, and in the case of a positive signal-to-noise ratio, at least one time constant is reduced. The time constant is preferably the switch-off time.The phrase "a time constant is reduced" preferably means that a faster time constant is set. Similarly, the phrase "a time constant is increased" preferably means that a slower time constant is set.

[0049] Alternatively or in addition to the frequency-dependent change of the time constant, a design is also possible in which another parameter of the dynamic scheme is modified depending on the frequency, e.g. a compression or expansion ratio or a respective knee point are changed independently of each other in different frequency ranges and therefore possibly differently.

[0050] The hearing system according to the invention comprises a signal processing unit configured to carry out a method as described above. Furthermore, the hearing system comprises a receiver for outputting a signal.

[0051] Preferably, the hearing system is designed as a hearing aid and includes one or more microphones for generating the input signal. The hearing aid is, for example, a BTE, ITE, or RIC device. The hearing aid has a housing in which the microphone is located. Preferably, the signal processing is also arranged in the housing. Advantageously, a battery is located in the housing to power the hearing aid. The receiver is either housed in the housing or connected to the housing via a lead, in particular a cable. The hearing aid is either monaural, with only a single housing, or binaural, with two housings, each containing one or more microphones and each containing a

[0052] The hearing aids have receivers or are connected to one. The housings of the binaural hearing aid are then worn on opposite sides of the user's head.

[0053] The presence of a microphone is not mandatory for the hearing system, so in a suitable design the hearing system does not have a microphone and is designed, for example, as a headphone.

[0054] An embodiment of the invention is explained in more detail below with reference to a drawing. The drawing schematically depicts: Fig. 1 a hearing system, Fig. 2 a block diagram of the hearing system, Fig. 3 a frequency-dependent modification of a parameter of a dynamic scheme.

[0055] Fig. 1 Figure 2 shows a hearing system 2, which in this case is designed as a hearing aid, more precisely as a RIC (receiver-in-canal) device, worn behind the ear by a user, particularly one with hearing loss. As a hearing aid, the hearing system 2, in this embodiment, has one or more, and in this case two, microphones 4 for generating an input signal E from sound signals in the environment. However, the presence of a microphone 4 is not mandatory, so in one variant the hearing system 2 has no microphone and is, for example, a headphone. Furthermore, the hearing system 2 has a signal processing unit 6 and a receiver 8, as well as a battery 10 for power supply. In this embodiment, the hearing system 2 also has a housing 12 in which the microphones 4, the signal processing unit 6, and the battery 10 are arranged. The receiver 8 is connected to the housing via a cable 14 and is in Fig. 1 integrated into an earpiece that is inserted into the user's ear.

[0056] During operation, the microphones 4 convert the ambient sound signals into an input signal E, which is fed to the signal processing unit 6 and modified by it. In this way, the signal processing unit 6 generates an output signal A, which is a modified input signal E. Here, the input signal E is amplified by a specific gain factor V. The output signal A, i.e., the amplified input signal E, is then output to the user via the earpiece 8. Both the input signal E and the output signal A are electrical signals. The earpiece 8 then converts the output signal A into an audible signal.

[0057] The hearing system 2 is either monaural or binaural. A binaural hearing system 2 then has, for example, two housings 12 with the associated components as shown in Fig. 1 shown, whereby the signal processing 6 does not have to be identical in both cases.

[0058] In Fig. 2 The hearing system 2 is shown as a block diagram. The signal processing unit 6 has an amplifier 16, to which the input signal E is fed. The amplifier then amplifies the signal and passes it on to the receiver 8 as the output signal A. The amplification in the amplifier 16 is achieved with a gain factor V, which is set by an automatic gain control. This control includes a dynamic processor 18. The hearing system 2 is operated in a program to which a specific dynamic scheme is assigned. The dynamic processor 18 operates according to this scheme during operation within that program. The respective dynamic scheme defines the gain factor V as a function of the level of the input signal E. This function is characterized by various parameters, for example...by one or more knee points, one or more compression or expansion ratios, an attack time, a release time, or a combination thereof.

[0059] The dynamics processor 18 is either a compressor or an expander, or a combination thereof. In an embodiment not shown, two dynamics processors 18 are present, one of which is a compressor and the other an expander, in which case the explanations for the illustrated embodiment apply analogously.

[0060] Additionally, a situation parameter S is determined, which characterizes the current acoustic environment and is then used during operation in one program to modify the assigned dynamic scheme. The dynamic processor 18 is then operated with a modified dynamic scheme depending on the situation parameter S. To determine the situation parameter S, a detector 20 is arranged, which here is part of the hearing system 2. The detector 20 is generally connected to the signal unit 6 and specifically to the automatic gain control, in this case, the dynamic processor 18.

[0061] Detector 20, for example, is a speech detector that determines whether speech is present in the environment. Alternatively, detector 20 determines the signal-to-noise ratio of the environment as a situation parameter S. For this purpose, in a suitable implementation, the input signal E is fed to detector 20 so that it determines the signal-to-noise ratio of the input signal E as a situation parameter S. Alternatively, detector 20 is a classifier that, for example, assigns the environment to a specific environmental class based on the input signal E. Alternatively, detector 20 is a motion detector, for example, an accelerometer, which determines a movement pattern as a situation parameter S. The motion detector determines, for example, whether the user is moving or not, how fast the user is moving, in which direction, or a combination thereof.Alternatively, detector 20 can be a stationarity detector, which determines the stationarity of the environment, i.e., the temporal dynamics of the ambient sound signals. For this purpose, detector 20 is supplied with, for example, the input signal E, whose stationarity, i.e., its temporal change, is then determined by detector 20 and output as a situation parameter S. Alternatively, detector 20 can be a diffuseness detector, which determines the diffuseness of the environment, i.e., the spatial distribution of the sound signals in the environment. For this purpose, detector 20 is supplied with, for example, the input signal E, whose directionality, i.e., its spatial level distribution, is then determined by detector 20 and output as a situation parameter S. Alternatively, detector 20 can be a rangefinder, which determines the distance of the hearing system 2 to a sound source in the environment and then outputs this distance as a situation parameter S.Alternatively, the detector 20 determines the reverberation time in the environment, e.g., based on an analysis of the input signal E. In a variant not shown, several identical or different detectors 20 mentioned above are combined.

[0062] In Fig. 3 Figure 1 shows an embodiment of a frequency-dependent modification of the dynamic scheme. In this embodiment, a parameter of the dynamic scheme is modified differently for different frequencies f, depending on the situation. The input signal E has a frequency spectrum which is divided into three frequency ranges B1-B3. Depending on the situation parameter S, the dynamic scheme is not only modified in the same way for the entire frequency spectrum, but specifically for the individual frequency ranges B1-B3. The previously described method of modifying the dynamic scheme is thus carried out in parallel in the frequency ranges B1-B3. In the embodiment shown, the switch-off time t_r of the dynamic scheme is changed depending on the frequency.A further distinction is made between speech frequency ranges B2, here from 800 Hz to 4 kHz, and speech-free frequency ranges B1 and B3, here below 800 Hz and above 4 kHz. In a variant not shown, the frequency spectrum is divided in more detail, in which the speech frequency ranges correspond to individual sounds or formants, and all remaining frequencies constitute speech-free frequency ranges.

[0063] The switch-off time t_r, and more generally a parameter of the dynamic scheme, is modified frequency-dependently by setting it on a respective frequency range B1-B3 depending on whether this frequency range B1-B3 is a speech frequency range B2 or a speech-free frequency range B1, B3. The frequency-dependent modification thus corresponds to a speech-dependent modification. Fig. 3 This is combined with a configuration in which the situation parameter S is a signal-to-noise ratio of the current acoustic environment. The frequency-dependent switch-off time t_r in the case of a positive signal-to-noise ratio is in Fig. 3Solid lines indicate the signal-to-noise ratio, while dashed lines indicate a negative signal-to-noise ratio. In the case of a negative signal-to-noise ratio, the switch-off time t_r is reduced on the speech frequency range B2, and conversely, in the case of a positive signal-to-noise ratio, the switch-off time t_r is increased. On the speech-free frequency ranges B1 and B3, the switch-off time t_r is increased in the case of a negative signal-to-noise ratio and reduced in the case of a positive signal-to-noise ratio. "Reduced" here means that a faster switch-off time t_r is set, and "increased" means that a slower switch-off time t_r is set. The same procedure is also generally suitable for the switch-on time.

[0064] Alternatively or in addition to the frequency-dependent change of the time constant, a design is also suitable in which another parameter of the dynamic scheme is modified depending on the frequency, e.g. a compression or expansion ratio or a respective knee point are changed independently of each other in different frequency ranges and therefore possibly differently. Reference symbol list

[0065] 2 Hearing system 4 Microphone 6 Signal processing 8 Receiver 10 Battery 12 Housing 14 Cable 16 Amplifier 18 Dynamic processor 20 Detector A Output signal B1, B3 Speech-free frequency range B2 Speech frequency range E Input signal S Situation parameter t_r Switch-off time V Gain factor

Claims

1. Method for operating a hearing aid system (2) in a current acoustic environment, - wherein the hearing aid system (2) comprises a signal processor (6), - wherein the signal processor (6) generates an output signal (A) from an input signal (E) by amplifying the input signal (E) by a gain factor (V), - wherein the hearing aid system (2) has a receiver (8), via which the output signal (A) is output, - wherein the gain factor (V) is set by an automatic gain control, which has a dynamic range processor (18) that can be operated with a dynamic range scheme which defines the gain factor (V) as a function of a level of the input signal (E), - wherein the hearing aid system (2) is operated in a program to which a dynamic range scheme is assigned, with which the dynamic range processor (18) is operated during operation in the program, - wherein at least one situation parameter (S) is determined, which characterizes the current acoustic environment, - wherein during operation in the program, the assigned dynamic range scheme is modified depending on the situation parameter (S), so that the dynamic range processor (18) is operated with a modified dynamic range scheme, characterized - in that the situation parameter (S) is a signal-to-noise ratio of the current acoustic environment, so that the dynamic range scheme is modified depending on the signal-to-noise ratio of the current acoustic environment, - in that the signal-to-noise ratio is a ratio of a signal which originates from a speaker and a noise which originates from one or more other sound sources, - in that the dynamic range scheme has two time constants, namely a switch-on time and a switch-off time, - in that in the case of a negative signal-to-noise ratio at least one of the time constants is reduced and conversely, in the case of a positive signal-to-noise ratio the at least one of the time constants is increased.

2. Method for operating a hearing aid system (2) in a current acoustic environment, - wherein the hearing aid system (2) comprises a signal processor (6), - wherein the signal processor (6) generates an output signal (A) from an input signal (E) by amplifying the input signal (E) by a gain factor (V), - wherein the hearing aid system (2) has a receiver (8), via which the output signal (A) is output, - wherein the gain factor (V) is set by an automatic gain control, which has a dynamic range processor (18) that can be operated with a dynamic range scheme which defines the gain factor (V) as a function of a level of the input signal (E), - wherein the hearing aid system (2) is operated in a program to which a dynamic range scheme is assigned, with which the dynamic range processor (18) is operated during operation in the program, - wherein at least one situation parameter (S) is determined, which characterizes the current acoustic environment, - wherein during operation in the program, the assigned dynamic range scheme is modified depending on the situation parameter (S), so that the dynamic range processor (18) is operated with a modified dynamic range scheme, characterized - in that a distinction is made between speech frequency bands (B2) and speech-free frequency bands (B1, B3), - in that a parameter of the dynamic range scheme is changed as a function of frequency by setting the parameter over a particular frequency band (B1, B2, B3) according to whether this frequency band (B1, B2, B3) is a speech frequency band (B2) or a speech-free frequency band (B1, B3), - in that the situation parameter (S) is a signal-to-noise ratio of the current acoustic environment, - in that the dynamic range scheme has two time constants, namely a switch-on time and a switch-off time, - in that in the case of a negative signal-to-noise ratio over a speech frequency band (B2) at least one of the time constants is reduced and in the case of a positive signal-to-noise ratio the at least one time constant is increased, - in that over a speech-free frequency band (B1, B3), in the case of a negative signal-to-noise ratio the at least one time constant is increased and in the case of a positive signal-to-noise ratio the at least one time constant is reduced.

3. Method according to Claim 2, wherein the dynamic range scheme is modified by changing a switch-on time of the dynamic range scheme or a switch-off time (t_r) of the dynamic range system, or both, as a function of frequency.

4. Method according to any one of Claims 1 to 3, wherein the dynamic range processor (18) is a compressor, so that the dynamic range scheme is a compression scheme.

5. Method according to any one of Claims 1 to 4, wherein the dynamic range processor (18) is an expander, so that the dynamic range scheme is an expansion scheme.

6. Method according to any one of Claims 1 to 5, - wherein the situation parameter (S) indicates whether speech is present in the environment or not, - wherein the dynamic range scheme is modified in favor of speech intelligibility if speech is present, and otherwise in favor of sound quality.

7. Method according to any one of Claims 1 to 6, - wherein the dynamic range scheme is modified by changing at least one parameter of the dynamic range scheme, - wherein the parameter is selected from a set of parameters comprising: a switch-on time, a switch-off time (t_r), a knee point, a compression ratio, an expansion ratio.

8. Method according to any one of Claims 1 to 7, - wherein the situation parameter (S) is an environment class, so that the dynamic range scheme is modified depending on the environment class, - wherein the environment class is in particular selected from a set of classes, comprising: speech, music, noise.

9. Method according to any one of Claims 1 to 8, - wherein the situation parameter (S) is a motion pattern, so that the dynamic range scheme is modified depending on a motion of the hearing aid system (2), - wherein the motion pattern is selected in particular from a set of motion patterns, comprising: a resting position, a walking motion, a running motion, driving.

10. Method according to any one of Claims 1 to 9, wherein the situation parameter (S) is a stationarity of the current acoustic environment and in particular of the input signal, so that the dynamic range scheme is modified depending on the stationarity of the current acoustic environment.

11. Method according to any one of Claims 1 to 10, wherein the situation parameter (S) is a diffuseness of the current acoustic environment, so that the dynamic range scheme is modified depending on the diffuseness of the current acoustic environment, wherein the diffuseness is a measure of a variation in the level in the environment.

12. Method according to any one of Claims 1 to 11, wherein the situation parameter (S) is a distance from the hearing aid system (2) to a sound source, in particular a speaker, in the current acoustic environment, so that the dynamic range scheme is modified depending on the distance.

13. Method according to any one of Claims 1 to 12, - wherein the situation parameter (S) is a reverberation time of the current acoustic environment, so that the dynamic range scheme is modified depending on the reverberation time, - wherein the dynamic range scheme has two time constants, namely a switch-on time and a switch-off time, - wherein the reverberation time is determined on a recurring basis and in the event of an increase in the reverberation time at least one of the time constants is increased and conversely reduced in the event of a decrease in the reverberation time.

14. Hearing aid system (2), comprising: - a signal processor (6), which is designed to carry out a method according to any one of Claims 1 to 13, - a receiver (8) for outputting an output signal (A).