Method for operating a hearing system and hearing system

Abstract

A method for operating a hearing system (2) is specified, wherein a dynamic approach is adapted for automatic gain adjustment in a situation-dependent manner. Furthermore, a corresponding hearing system (2) is specified.

Description

Technical Field

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

[0002] In the most general design, a hearing system has a signal processing device and an earpiece, wherein the earpiece is also referred to as a loudspeaker. An input signal is fed to the signal processing device and then modified by the signal processing device to generate an output signal, that is, the output signal is a modified input signal. For example, the input signal is amplified with a specific amplification factor. In the following, the input signal and the output signal are also referred to as signals. Finally, the output signal is output by means of the earpiece. The input signal and the output signal are electrical signals. The earpiece then converts the output signal into an acoustic signal.

[0003] Optionally, the hearing system additionally has a microphone, which generates an input signal by receiving one or more sound signals from the environment and converting them into an input signal. Such a hearing system with an additional microphone is also called a hearing device, which amplifies the sound signal from the environment by using the hearing device and outputs it to the hearing-impaired person as an amplified sound signal for providing the hearing-impaired person.

[0004] In hearing systems, instead of simple linear amplification, nonlinear amplification can be performed. This is achieved, for example, with so-called compressors, which provide and apply a compression scheme which sets the amplification factors in a suitable manner when modifying the input signal, that is, when generating the output signal. The behavior of the compressor, i.e. the compression scheme, is defined by one or more parameters, more precisely compression parameters.

[0005] US 7,773,763 B2 describes a hearing device in which different programs are associated with different environment types. Each program contains specific values ​​for specific parameters of the hearing device in order to achieve the best signal quality in a given environment. The hearing device can classify the current environment, associate the current environment with the environment type, and then select a suitable program with the correspondingly optimal parameters.

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

[0007] DE 10 2010 041 740 A1 describes a method in which different dynamic compressions are performed for different frequency bands. Summary of the invention

[0008] Against this background, the technical problem to be solved by the present invention is to provide an improved method for operating a hearing system and a corresponding hearing system, in particular to improve the dynamics in the hearing system by means of automatic gain control.

[0009] According to the present invention, the above technical problem is solved by a method having the features according to the present invention and by a hearing system having the features according to the present invention. Advantageous designs, extensions and variants are the subject of the following description. The description of the method also applies to the hearing system, and vice versa.

[0010] The method is used for operating a hearing system and is therefore an operating method. During operation, the hearing system is worn by a user, in particular on, in or behind the ear, that is to say generally on the body and in the region of one or both ears. At a given point in time, the hearing system and the user are in a current acoustic environment. This is understood to mean that the environment around the user and the hearing system at the current point in time forms a specific acoustic scene.

[0011] The hearing system has a signal processing device, which generates an output signal of the hearing system by amplifying the input signal of the hearing system with an amplification factor according to the input signal of the hearing system. In other words, the signal processing device is used as an amplifier. The signal processing device is preferably integrated in the hearing system. Preferably, the hearing system is a hearing device, particularly preferably a hearing device for supplying hearing-impaired persons. As a hearing device, the hearing system has at least one microphone, which receives an acoustic signal, i.e. a sound signal, from the environment and converts it into an input signal. The hearing system also has an earpiece, via which the output signal, i.e. the amplified input signal, is output. The earpiece is also called a loudspeaker.

[0012] The amplification factor is set by means of an automatic gain control, which is in particular part of the signal processing device. Automatic gain control is also referred to as AGC (automatic gain control). "Automatic" is understood in particular to mean that the gain adjustment is performed without user input, but rather automatically during operation. In addition, for the automatic gain adjustment, in a suitable embodiment, the hearing system also has a manual gain control, via which the user can additionally influence the amplification factor according to his or her preferences.

[0013] The automatic gain control has a dynamic processor which can operate with a dynamic scheme which defines the amplification factor as a function of the level of the input signal, i.e. the input level. In other words, the dynamic scheme specifies which amplification factor is used at which input level. The dynamic scheme is also called a dynamic algorithm. Generally speaking, examples of dynamic schemes are compression or expansion schemes, in particular AGCi (automatic gain control input dependent) and ADRO (adaptive dynamic range optimization). Various dynamic methods are suitable here, with AGCi being particularly preferred.

[0014] Dynamic processors are now characterized in that the amplification factor is not constant, but rather the dynamic processor causes a nonlinear amplification. The dynamic processor adjusts the gain, in particular by means of a signal processing device, as a function of a control variable. As a control variable, for example, the input level is used, or alternatively, the output level is used. Both variants have specific advantages and disadvantages. The corresponding level is measured, for example, by means of a level detector, which is preferably part of the hearing system.

[0015] Generally speaking, the input signal and the output signal each have a dynamic, i.e. a level in a specific range extending from a minimum level 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 is correspondingly referred to as the output level. Similarly, the dynamics of the input signal are referred to as the input dynamics, and the dynamics of the output signal are referred to as the output dynamics. The dynamics of the output signal are related to the dynamics of the input signal and the modifications made by the signal processing device. In specific application cases, it is desirable to limit, i.e. compress or increase, i.e. expand the dynamics of the output signal. In a hearing device for hearing-impaired persons, it is appropriate to match the dynamics to the hearing of the hearing-impaired person, which is reduced relative to that of a person with normal hearing. The reduced hearing is limited to a specific level range, which then appropriately defines the dynamics of the output signal. The dynamics of the output signal are then mapped to this level range by generating an output signal with correspondingly suitable dynamics, in particular to achieve the highest sound quality. Then, according to the design, it is advantageous to perform compression or expansion or even a combination thereof.

[0016] At runtime, the hearing system is run with a specific program associated with a dynamic scheme. That is, the dynamic processor is set up in a predefined way, i.e. preconfigured, by the program. The program and the dynamic scheme are adapted to the current acoustic environment so as to produce an optimal output, i.e. output quality. That is, during runtime, the dynamic processor is run with a program with an associated dynamic scheme.

[0017] Now, the core idea of ​​the present invention is, in particular, to determine at least one situation parameter, which characterizes the current acoustic environment, in particular quantitatively or qualitatively. Here, during the operation of a program, the associated dynamic scheme is modified according to the situation parameter, so that the dynamic processor is operated using the modified dynamic scheme. That is, the dynamic scheme pre-given by the program for the current acoustic environment is optimized according to the situation. In other words: the dynamic scheme for automatic gain adjustment is modified according to the situation. This is based on the consideration that the acoustic environment may change, so that the selected dynamic scheme is no longer optimal, wherein, but in some cases, the amplitude or duration of the change is not enough to justify the rationality of changing the program, for example, from a speech program to a music program. This advantageously avoids sudden changes in the dynamic scheme, on the contrary, the dynamic scheme that is actually associated with the environment is adjusted according to the situation, that is, it can be said that it is further developed or further formed in the process of fine tuning. As a result, the output of the hearing system is improved in a comfortable way for the user.

[0018] In particular, the dynamics are not adjusted solely as a function of the input level or output level, but rather situation parameters are preferably additionally fed to the automatic gain unit and used as additional control or regulating variables.

[0019] The situation parameter is preferably determined by means of a detector. The detector is preferably part of the hearing system. In an alternative, the situation parameter is determined externally with respect to the hearing system and transmitted to the hearing system, for example via a data connection.

[0020] The following design is particularly preferred, in which the dynamic processor is a compressor, so that the dynamic scheme is a compression scheme. The following design is also advantageous, in which the dynamic processor is an expander, so that the dynamic scheme is an expansion scheme. Compressors and expanders generally work in opposite ways. A compressor limits the amplification factor above a threshold value, while an expander generally increases the amplification factor above a threshold value. In other words: a compressor amplifies a signal with a low level more than a signal with a relatively high level, and conversely, an expander amplifies a signal with a high level more than a signal with a relatively low level. In summary, therefore, a compressor is used to reduce the dynamics of a signal, and a compressor is used to increase the dynamics of a signal.

[0021] The following design scheme is also advantageous, in which the compressor and the expander are combined with each other either as two separate dynamic processors or together as one dynamic processor. In a first design scheme, the compressor and the expander are used independently of each other, so that only one of the two is always active, for example, depending on the situation parameters or the program activity of the hearing system set. In contrast, in a second design scheme, the compressor and the expander are simultaneously active in a so-called mixed mode, preferably in such a way that the compressor generally reduces the level, while the expander increases the dynamics again at the working point, i.e., within a limited frequency range and / or level range. In other words: using the expander to locally inefficiently compress is itself global. In principle, the opposite design scheme is also possible and appropriate, in which the expander similarly acts globally, while the compressor only acts locally.

[0022] In a given situation, it is technically possible in principle to use either a compressor or an expander or a combination of both. However, which of the three previously mentioned design solutions makes the most sense from an audiological point of view depends on the situation. In particular, although the combination of a compressor and an expander also initially appears contradictory, it is technically possible in principle and makes sense in certain situations.

[0023] In the following, without limiting the generality, it is assumed that a compressor or an expander is used. However, these descriptions are also similarly applicable to the following design schemes, in which not only a compressor but also an expander is used, and according to the specific design scheme, multiple dynamic schemes may be obtained, and then the described preferred design schemes are appropriately applied to these dynamic schemes respectively.

[0024] Modifications within a given program are ultimately special adjustments to the program itself for individual deviations of the current acoustic environment from the idealized environment or standard environment predicted when the program was created. The problem when a hearing system is running is that different situations exist simultaneously in different environments, which require mutually exclusive programs. A particularly important situation is the presence of speech in the environment. In order to make these speech as understandable as possible to the user, it is appropriate to set up a program that improves speech understanding. Here, the true reproduction of other sound signals or noise is secondary, on the contrary, the speech should first be recognizable to the user. For this purpose, a special dynamic scheme designed for maximum speech understanding is particularly set and used. On the contrary, in an environment without speech, it is appropriate to strive to reproduce the acoustic environment as realistically as possible, that is, the best possible sound quality should be achieved. The best possible sound quality is particularly understood as compensating the user's hearing loss as best as possible, that is, compensating for hearing loss to the greatest extent. This is particularly important for music, which is severely distorted by dynamic schemes in some cases to improve speech intelligibility. The same applies to other sound signals in the environment, which are sometimes distorted to the extent that the user can no longer recognize and can no longer associate. This leads to particular problems when not only speech but also other sound signals are present in the environment, that is, when the acoustic environment generally includes a plurality of different sound signals, which are associated with different programs for optimal output, for example, a speech program on the one hand and a music program on the other hand. In principle, a compromise can be made here in that a program is used that is not only optimized in one direction, but rather an attempt is made to reproduce all sound signals at least partially in a suitable manner. However, it is obvious here that no sound source is then output in an optimal manner, that is, with such a program, speech can never be optimally understood.

[0025] This problem is circumvented by fine-tuning the dynamics scheme within the program. The dynamics scheme is modified in a situation-dependent manner by determining at least one situation parameter and, in particular, by repeatedly determining a plurality of situation parameters in order to ensure an optimal setting of the dynamics processor at a given time. The dynamics processor is thus adjusted in a situation-specific manner. In a suitable embodiment, the situation parameters are determined by the hearing system itself.

[0026] In particular, the target conflict between speech understanding and sound quality described above can be advantageously resolved thereby. For this reason, in a particularly advantageous design, the situation parameter gives whether there is speech in the environment, and if there is speech, the dynamic scheme is modified in a manner that is conducive to speech understanding, otherwise, the dynamic scheme is modified in a manner that is conducive to sound quality. That is to say, from the beginning, a program optimized in terms of the highest sound quality is particularly set, advantageously a music program. Then, in the case of speech, speech understanding is prioritized, and the program is modified in a manner that obtains maximum speech understanding, particularly without changing the actual program, specifically modifying the dynamic scheme. Here, it is suitable to keep modifying only when there is speech. Here, the modification is performed automatically, advantageously without user input. The user also does not need to select a specific speech program or a sound or music program. Therefore, the current acoustic environment is reacted directly, and there is no need to detour via program selection. That is to say, the program is not static, but is dynamic due to modification based on the situation parameter. It is thus guaranteed that the user experiences the highest sound quality in principle, but in the case of speech, experiences the maximum speech understanding. In order to determine whether there is speech, the suitable detector is a speech detector, which checks the input signal in particular for whether there is speech.

[0027] The dynamic scheme defines the behavior of the dynamics processor in operation and is defined by one or more parameters, more precisely by dynamic parameters, such as compression parameters and / or expansion parameters. Suitable dynamic parameters are in particular the compression or expansion threshold (also called inflection point), the compression or expansion ratio (also simply called compression or expansion), the opening time (also called attack) and the closing time (also called release). The opening time and closing time of the dynamics processor are correspondingly also called time constants. That is to say, the dynamic scheme is preferably modified by changing at least one parameter of the dynamic scheme, wherein the parameter is selected from a set of parameters, the set of parameters comprising: opening time, closing time, inflection point, compression ratio in the case of a compressor, expansion ratio in the case of an expander. The following design is particularly preferred, in which the opening time or the closing time is changed in a situation-dependent manner. However, the remaining parameters are also suitable for modifying the dynamic scheme.

[0028] Furthermore, in this case, in particular, no switch is made between two different dynamic schemes in the program, but rather the same dynamic scheme, i.e. the same dynamic algorithm, is used in principle, but its parameters are adjusted in a situation-dependent manner, so that the basic design on which the dynamic scheme is based is retained and only the basic design is modified with regard to its specific implementation. This is in contrast to DE 10 2005 061 000 A1 mentioned at the outset, in which a switch is made between different dynamic schemes in principle, depending on the situation. In this case, in contrast, the programming of the dynamic scheme is dynamically changed.

[0029] The dynamic range for maximizing sound quality and speech intelligibility is known in principle. For the highest sound quality, for example, an amplification that is as linear as possible in terms of the result is sought, so that the dynamic scheme first compresses high input levels and does not compress low and intermediate input levels, for which purpose the compression ratio is in particular 1. The inflection point marking the transition to a greater compression ratio is then arranged relatively high, that is to say compared to the inflection point of the dynamic scheme for maximum speech intelligibility. Here, for example, the level range of the input signal containing speech is mapped as far as possible to the level range that can be heard by the user.

[0030] In an advantageous design, the situation parameter is the current acoustic environment, in particular the signal-to-noise ratio of the input signal, so that the dynamic scheme is modified according to the signal-to-noise ratio of the current acoustic environment. That is, the dynamic scheme is set according to the relative strength of the interfering noise relative to the useful signal, i.e. the sound signal, that the user in the environment wants to perceive. This is based on the following considerations, namely, in the case of a high signal-to-noise ratio, the useful signal is already clear enough for the user, and then for the sake of sound quality, only small adjustments, in particular compression, are required, and small adjustments, in particular compression, are performed accordingly. On the contrary, in the case of a low signal-to-noise ratio, there is an environment with strong interference, so that the dynamic scheme is modified in a manner specifically adjusted for this, so as to enhance the useful signal relative to the interfering signal, thereby making it clear for the user. As a detector for the signal-to-noise ratio, it is appropriate to use a noise detector, in particular to feed the input signal to the noise detector, and then determine the signal-to-noise ratio of the environment based on the input signal.

[0031] In another advantageous design, the signal-to-noise ratio is the ratio of the signal originating from the speaker to the noise originating from one or more other sound sources in the environment. Other sound sources are, for example, other people, such as listeners or devices, such as the vehicle in which the user and the speaker are located. Here, the dynamic scheme has two time constants, namely the opening time and the closing time. Now, in the case of a negative signal-to-noise ratio, at least one of the time constants is reduced, and in the case of a positive signal-to-noise ratio, at least one of the time constants is increased. "Reducing the time constant" is preferably understood as setting a fast time constant. "Increasing the time constant" is correspondingly preferably understood as setting a slow time constant. Generally speaking, for the opening time, "fast" is understood as a time of up to 10ms, and "slow" is understood as a time of 100ms to 1s. For the closing time, generally speaking, "fast" is understood as a time of up to 100ms, and "slow" is understood as a time of 500ms to 1s.

[0032] Preferably, when the signal-to-noise ratio is negative, the two time constants are reduced, whereas when the signal-to-noise ratio is positive, the two time constants are also increased. This is based on the consideration that an environment with a negative signal-to-noise ratio is very laborious for the user. In this case, the user particularly benefits from the gaps in the input signal, i.e., from short time segments with very little noise, wherein "short" is particularly understood as a length of up to one second. It is appropriate to amplify these gaps by modifying the dynamic scheme with a correspondingly fast time constant in accordance with the situation. In this context, "fast" is particularly understood as a time constant of a maximum of 100ms. In the case of strong noise, this method improves speech intelligibility, but leads to a deterioration in sound quality, so in the case of correspondingly small noise, this method is abandoned and a relatively long time constant, particularly a time constant longer than 100ms, is selected instead, so that the adjustment, specifically, the compression is generally weaker.

[0033] In another advantageous design, the situation parameter is an environment category, so that the dynamic scheme is modified according to the environment category, wherein the environment category is selected in particular from a set of categories, which categories include: speech, music, interference noise. Here, the situation parameter is determined in particular with the help of a detector constructed as a classifier. The classifier is preferably part of a signal processing device and determines the environment category in particular based on the input signal. If the environment category is music, the dynamic scheme is conservatively modified, that is, for example, when compressing, for example, the compression is smaller than in the case of speech. Then, in the case of speech, the dynamic scheme is modified for maximum speech intelligibility. This is advantageously used in concerts or similar environments where there is no speech. Then, the dynamic scheme is modified for the most realistic reproduction of music, rhythm and melody, that is, in particular, strive to amplify as linearly as possible for the user in the result.

[0034] In another advantageous design, the situation parameter is a movement pattern, so that the dynamic scheme is modified according to the movement of the hearing system, wherein the movement pattern is selected in particular from a set of movement patterns including: static position, walking movement, running movement, driving. In particular, the movement pattern is determined by means of a detector, which is constructed as a motion sensor, preferably by means of an acceleration sensor. In this case, in particular, a distinction is made between movement and stillness. It is suitable to further divide the movement into different types of movement, for example according to speed. For example, it is recognized whether the user is stationary, moving, or driving in a car. This is based on the idea that, in general, the importance of speech intelligibility is different in different environmental situations, and therefore it is suitable to modify the dynamic scheme according to the user's movement state or movement behavior.

[0035] In another advantageous design, the situation parameter is the stability of the current acoustic environment, in particular the stability of the input signal, so that the dynamic scheme is modified according to the stability of the current acoustic environment. In order to determine the stability, it is suitable to use a stability detector, in particular to feed the input signal to the stability detector, and then determine the stability based on the input signal. Stability is in particular a measure of the dynamics of one or more sound sources in the environment, i.e. a measure of the change of the level of this sound source or these sound sources over time. In other words, stability gives the dynamics of the level related to time.

[0036] In another advantageous design, the situation parameter is the diffuseness of the current acoustic environment, so that the dynamic scheme is modified according to the diffuseness of the current acoustic environment. In order to determine the diffuseness, it is appropriate to use a diffuseness detector, in particular to feed the input signal to the diffuseness detector, and then determine the diffuseness based on the input signal. Diffusivity is similar to the stability described above, because diffuseness is a measure of the change of the level in the environment, but now it is not time-related, but position-related. In other words, diffuseness is how the directionality of the environment is given, that is, whether the sound signal comes from a specific direction towards the user, that is, it is directional, or uniformly from all directions, that is, it is diffuse. In other words, diffuseness gives the dynamics of the level related to the direction. That is, a very uniform level in space leads to a high degree of diffusion.

[0037] In another advantageous embodiment, the situation parameter is the distance between the hearing system, in particular a possible microphone, and a sound source in the current acoustic environment, in particular a speaker, so that the dynamic solution is modified according to the distance. In particular, the distance is determined by means of a distance meter 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 sound sources that are farther away are generally more difficult to understand, so that in this case it is appropriate to modify the dynamic solution for a higher speech intelligibility.

[0038] In a further advantageous embodiment, the situation parameter is the reverberation time of the current acoustic environment, so that the dynamic scheme is modified as a function of the reverberation time, wherein the dynamic scheme has two time constants, namely a start time and a stop time, wherein the reverberation time is determined repeatedly and at least one of the time constants is increased in the case of an increase in the reverberation time, and conversely at least one of the time constants is reduced in the case of a decrease in the reverberation time. This is based on the idea that in environments with strong echoes, fast adjustments, in particular fast compressions, i.e. short time constants in general, are quite disadvantageous. The reverberation time is advantageously determined by means of a detector, to which the input signal is fed, so that the detector determines the reverberation time based on an analysis of the input signal.

[0039] The combination of the different designs described above is also particularly advantageous. Then, in a suitable design, a plurality of situation parameters are determined and used to control the automatic gain adjustment. In a suitable design, the same situation parameters are redundantly determined by means of a plurality of identical or different detectors.

[0040] In another advantageous design, the one program is a universal program. The universal program is characterized in particular in that it is not adapted to a specific predefined situation or used within a specific limited application range, but is used in a plurality of different situations. In a suitable design, the hearing system is operated with the aid of only one program, that is, a design with different programs is abandoned, which are respectively specially adjusted and optimized for different environments. Instead, the adaptation of the pre-modified acoustic environment is performed only by dynamic schemes modified in a situation-dependent manner. In an equally suitable design, the hearing device has one or more special programs respectively adjusted for special, specific situations in addition to the universal program. Preferably, the corresponding special program can be set by the user and preferably not automatically set. Special programs are immutable, that is, for each specific situation, specifically for the dynamic scheme, fixed, predetermined parameter values ​​are generally pre-given, while universal programs are variable at least based on dynamic schemes adapted to the situation. Therefore, universal programs are also generally referred to as automatic programs, because they are automatically adapted to the environment, while in contrast, individual special programs are static. In the case of the universal program, the situation parameters are in particular fed directly to the automatic gain control and used to control the amplification factor. Here, the possibility of manual gain setting is preferably also abandoned. Manual adjustments by the user are expediently only possible by selecting a special program. Starting from the universal program, all adjustments are automatically made in a situation-dependent manner by modifying the dynamic scheme according to the situation parameters.

[0041] In a preferred design, the dynamic scheme is modified in a frequency-dependent manner. This is particularly understood to be that one or more parameters of the dynamic scheme are changed in different ways for different frequencies in a situation-dependent manner. The input signal has a spectrum that is now divided into at least two frequency ranges. A frequency range is particularly a single frequency or a frequency interval. It is suitable to divide the input signal into a plurality of continuous frequency ranges by means of a filter bank of a signal processing device. Except for a slight overlap in the edge range that may be due to technical reasons, these frequency ranges are preferably non-overlapping. Now, the dynamic scheme is not adjusted in the same way for the entire spectrum only according to the situation parameters, but the dynamic range is specifically adjusted for each frequency range. Therefore, it can be said that the method for modifying the dynamic scheme described above is performed in parallel in multiple frequency ranges. In particular, further improved compression or expansion can be achieved thereby. It is particularly advantageous to modify the dynamic scheme in a targeted manner for a specific frequency range, thereby ensuring the best match with the current acoustic environment. The frequency-dependent modification of the dynamic scheme is suitable for all situation parameters described. In principle, designs are also suitable in which compression or expansion or a combination of both is performed in different frequency ranges as required.

[0042] Therefore, in the frequency-dependent modification, the parameters that are changed in order to modify the dynamic scheme are changed differently in a situation-dependent manner in different frequency ranges. The following design is particularly suitable, in which the dynamic scheme is modified by changing the start time of the dynamic scheme or the stop time of the dynamic scheme or both in a frequency-dependent manner. In other words: the start time or the stop time or both parameters are changed independently of each other for different frequency ranges, so that different time constants are obtained for different frequency ranges if necessary. As a result, the time constant is modified specifically for a specific frequency range, without performing the same modification for all other frequency ranges.

[0043] In a preferred design, a distinction is made between a speech frequency range and a frequency range without speech, that is, between a frequency range containing speech and a frequency range without speech. A suitable speech frequency range is a frequency range from 800 Hz to 4 kHz or a subset thereof. Correspondingly, a suitable frequency range without speech is a frequency range below 800 Hz and above 4 kHz. Alternatively, a more detailed division of the spectrum is performed, wherein the speech frequency range corresponds to individual phonemes (Lauten) or formants, and all the remaining frequencies then form a frequency range without speech. Preferably, the parameters of the dynamic scheme are changed in a frequency-dependent manner by setting parameters for the frequency range according to whether the corresponding frequency range is a speech frequency range or a frequency range without speech. Therefore, the frequency-dependent modification corresponds to the speech-dependent modification. In short, the speech frequency range is advantageously distinguished from the frequency range without speech, and the dynamic scheme is modified accordingly and optimally for these two types of frequency ranges. Thus, improved speech intelligibility is particularly obtained. The speech-dependent modification of the dynamic scheme is also suitable for all the described situation parameters.

[0044] The combination of speech-related modifications with the already mentioned design scheme is particularly advantageous in which the situation parameter is the signal-to-noise ratio of the current acoustic environment, in particular the signal-to-noise ratio of the input signal. For the speech frequency range, in the case of a negative signal-to-noise ratio, as already described, at least one of the time constants of the dynamic scheme is reduced, and conversely, in the case of a positive signal-to-noise ratio, at least one time constant is increased. Then, here, in contrast, for the frequency range without speech, 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 a stop time. "Reducing the time constant" is preferably understood as setting a fast time constant. "Increasing the time constant" is correspondingly preferably understood as setting a slow time constant.

[0045] As an alternative or in addition to changing the time constant in a frequency-dependent manner, a design concept is also suitable in which another parameter of the dynamic concept is modified in a frequency-dependent manner, for example, independently of each other, thereby changing the compression ratio or expansion ratio or the corresponding inflection points in different frequency ranges to different degrees if necessary.

[0046] The hearing system according to the invention has a signal processing device which is designed to carry out the method described above. The hearing system also has a receiver which is used to output the output signal.

[0047] Preferably, the hearing system is constructed as a hearing device and has one or more microphones for generating input signals. The hearing device is, for example, a BTE, ITE or RIC device. The hearing device has a housing for accommodating the microphone. Preferably, the signal processing device is also arranged in the housing. A suitable battery is arranged in the housing for providing energy for the hearing device. The earpiece is either accommodated in the housing or connected to the housing via a feeder, in particular a cable. The hearing device is either constructed as a monaural device with only one housing, or as a binaural device with two housings, each of which has one or more microphones and each has an earpiece or is connected to the earpiece. The housings of the binaural hearing device are then worn on different sides of the user's head.

[0048] The presence of a microphone in the hearing system is not mandatory, so that in one suitable embodiment the hearing system has no microphone and is designed, for example, as a headset. BRIEF DESCRIPTION OF THE DRAWINGS

[0049] The embodiments of the present invention are described in detail below with reference to the accompanying drawings.

[0050] Figure 1 The hearing system is schematically shown accordingly,

[0051] Figure 2 A block diagram of a hearing system is schematically shown accordingly,

[0052] Figure 3 The frequency-dependent modification of the parameters of the dynamic scheme is correspondingly schematically illustrated. DETAILED DESCRIPTION

[0053] Figure 1 A hearing system 2 is shown, which is designed as a hearing device, more precisely as an RIC device, which is worn behind the ear by a user, in particular a hearing-impaired user. As a hearing device, the hearing system 2 in this embodiment has one or more, here two, microphones 4 for generating an input signal E based on sound signals from the environment. However, the presence of the microphone 4 is not mandatory, so in a variant the hearing system 2 has no microphone and the hearing system 2 is, for example, a headset. Furthermore, the hearing system 2 has a signal processing device 6 and an earpiece 8 as well as a battery 10 for energy supply. Furthermore, here the hearing system 2 has a housing 12, in which the microphone 4, the signal processing device 6 and the battery 10 are arranged. The earpiece 8 is connected to the housing via a power supply line 14 and is connected to the housing 12 via a power supply line 14. Figure 1 The device is integrated into an earbud that is inserted into the user's ear.

[0054] In operation, the microphone 4 converts the sound signal in the environment into an input signal E, feeds the input signal E to the signal processing device 6 and is modified by the signal processing device 6. In this way, the output signal A is generated by the signal processing device 6, that is, the output signal A is the modified input signal E. Here, the input signal E is amplified with a specific amplification factor V. Finally, the output signal A, that is, the amplified input signal E, is output to the user by means of the earpiece 8. The input signal E and the output signal A are electrical signals respectively. Then, the earpiece 8 converts the output signal A into an acoustic signal.

[0055] The hearing system 2 is designed to be monaural or binaural. Then, the binaural hearing system 2 has, for example, two housings 12, which have Figure 1 The relevant components shown, among which the signal processing device 6 , are not necessarily identical in both cases.

[0056] exist Figure 2 In the figure, a hearing system 2 is shown as a block diagram. The signal processing device 6 has an amplifier 16, to which an input signal E is fed, which then amplifies the input signal E and forwards it to the earpiece 8 as an output signal A. The amplification in the amplifier 16 takes place with an amplification factor V, which is set by an automatic gain control. For this purpose, the automatic gain control has a dynamics processor 18. The hearing system 2 is now operated in a program to which a specific dynamic scheme is assigned, and the dynamics processor 18 is operated in this program with this specific dynamic scheme during operation. The corresponding dynamic scheme defines the amplification factor V as a function of the level of the input signal E. This function is characterized by various parameters, here for example by one or more inflection points, one or more compression or expansion ratios, opening time (attack), closing time (release) or a combination thereof.

[0057] The dynamics processor 18 is here a compressor or an expander or a combination thereof. In an embodiment not shown, there are two dynamics processors 18, one of which is a compressor and the other is an expander, wherein the description of the embodiment shown then applies analogously.

[0058] In addition, a situation parameter S characterizing the current acoustic environment is determined, which is then used during operation with the one program to modify the associated dynamic scheme. The dynamics processor 18 then operates with the modified dynamic scheme as a function of the situation parameter S. For determining the situation parameter S, a detector 20 is arranged, which is part of the hearing system 2 here. The detector 20 is generally connected to the signal unit 6, in particular to the automatic gain adjustment, and here in particular to the dynamics processor 18.

[0059] The detector 20 is, for example, a speech detector, which determines whether speech is present in the environment. Alternatively, the detector 20 determines the signal-to-noise ratio in the environment as a situation parameter S. To this end, in a suitable embodiment, the input signal E is fed to the detector 20, so that the detector 20 determines the signal-to-noise ratio of the input signal E as the situation parameter S. Alternatively, the detector 20 is a classifier, which, for example, associates the environment with a specific environment category based on the input signal E. Alternatively, the detector 20 is a motion detector, for example, an acceleration sensor, which determines the motion pattern as the situation parameter S. For example, the motion detector determines whether the user is moving, or how fast or in which direction the user moves, or a combination thereof. Alternatively, the detector 20 is a stability detector, which determines the stability of the environment, i.e., the dynamics of the sound signal of the environment over time. To this end, for example, the input signal E is fed to the detector 20, and then the detector 20 determines the stability of the input signal E, i.e., its change over time, and outputs it as the situation parameter S. Alternatively, the detector 20 is a diffuseness detector, which determines the diffuseness of the environment, i.e., the spatial distribution of the sound signal in the environment, for example. For this purpose, for example, an input signal E is fed to the detector 20, which then determines the directivity of the input signal E, i.e. its horizontal distribution in space, and outputs it as a situation parameter S. Alternatively, the detector 20 is a distance meter, which determines the distance between the hearing system 2 and 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, for example by means of an analysis of the input signal E. In a variant that is not shown, a plurality of identical or different detectors 20 mentioned above are combined.

[0060] exist Figure 3 In the embodiment of the frequency-dependent modification of the dynamic scheme is shown. Here, the parameters of the dynamic scheme are changed in different ways for different frequencies f in a situation-dependent manner. The input signal E has a spectrum that is divided into three frequency ranges B1-B3 here. Now, the dynamic scheme is not adjusted in the same way for the entire spectrum according to the situation parameter S, but the dynamic range is adjusted specifically for each frequency range B1-B3. Therefore, it can be said that the method for modifying the dynamic scheme described above is performed in parallel in the frequency range B1-B3. In the embodiment shown, the stop time t_r of the dynamic scheme is changed in a frequency-dependent manner. Here, additionally, a distinction is made between a speech frequency range B2 (here from 800Hz to 4kHz) and a frequency range B1, B3 (here below 800Hz and above 4kHz) without speech. In a variant that is not shown, the spectrum is divided in more detail, wherein the speech frequency range corresponds to each phoneme or formant, and all the remaining frequencies then form a frequency range without speech.

[0061] Here, the stopping time t_r, or in general the parameters of the dynamic scheme, are changed in a frequency-dependent manner by setting parameters for the frequency range B1-B3, depending on whether the corresponding frequency range B1-B3 is a speech frequency range B2 or a frequency range B1, B3 without speech. Thus, a frequency-dependent modification corresponds to a speech-dependent modification. Figure 3 In this, this is combined with a design in which the situation parameter S is the signal-to-noise ratio of the current acoustic environment. Figure 3 , the frequency-dependent stop time t_r in the case of a positive signal-to-noise ratio is shown by a solid line, and the frequency-dependent stop time t_r in the case of a negative signal-to-noise ratio is shown by a dotted line. For the speech frequency range B2, when the signal-to-noise ratio is negative, the stop time t_r is reduced, and conversely, when the signal-to-noise ratio is positive, the stop time t_r is increased. Then, conversely, for the frequency ranges B1 and B3 without speech, when the signal-to-noise ratio is negative, the stop time t_r is increased, and when the signal-to-noise ratio is positive, the stop time t_r is reduced. Here, "reducing" is understood as setting a fast stop time t_r, and "increasing" is understood as setting a slow stop time t_r. But in principle the same method also applies to the start time.

[0062] As an alternative or in addition to changing the time constant in a frequency-dependent manner, a design concept is also suitable in which another parameter of the dynamic concept is modified in a frequency-dependent manner, for example, independently of each other, thereby changing the compression ratio or expansion ratio or the corresponding inflection points in different frequency ranges to different degrees if necessary.

[0063] Reference numerals list

[0064] 2 Hearing System

[0065] 4 Microphones

[0066] 6 Signal processing device

[0067] 8. Earpiece

[0068] 10. Battery

[0069] 12 Housing

[0070] 14 Feeder

[0071] 16 Amplifier

[0072] 18 Dynamic Processors

[0073] 20 Detector

[0074] A Output signal

[0075] B1, B3 have no frequency range of speech

[0076] B2 Speech frequency range

[0077] E Input signal

[0078] S-case parameters

[0079] t_r stop time

[0080] V Amplification Factor

Claims

1. A method for operating a hearing system (2) in a current acoustic environment, in, The hearing system (2) has a signal processing device (6), The signal processing device (6) generates an output signal (A) from the input signal (E) by amplifying the input signal (E) with an amplification factor (V). The hearing system (2) has an earpiece (8), and the output signal (A) is output via the earpiece (8). wherein the amplification factor (V) is set by automatic gain adjustment with a dynamics processor (18), the dynamics processor (18) being capable of operating with a dynamic scheme defining the amplification factor (V) as a function of the level of the input signal (E), wherein the hearing system (2) is operated in a program, the program being associated with a dynamic scheme, with which the dynamic processor (18) is operated during operation in the program, wherein at least one situation parameter (S) is determined, the situation parameter characterizing the current acoustic environment, wherein, during the operation of the one program, the associated dynamic scheme is modified according to the situation parameter (S), so that the dynamic processor (18) is operated using the modified dynamic scheme, It is characterized in that The situation parameter (S) is the signal-to-noise ratio of the current acoustic environment, so that the dynamic scheme is modified according to the signal-to-noise ratio of the current acoustic environment. The signal-to-noise ratio is the ratio of the signal originating from the speaker to the noise originating from one or more other sound sources. The dynamic scheme has two time constants, the start time and the stop time. In case of a negative signal-to-noise ratio, at least one of the time constants is reduced, whereas in case of a positive signal-to-noise ratio, at least one of the time constants is increased.

2. The method according to claim 1, in, The dynamics processor (18) is a compressor, whereby the dynamics scheme is a compression scheme.

3. The method according to claim 1, in, The dynamics processor (18) is an expander, so that the dynamics scheme is an expansion scheme.

4. The method according to any one of claims 1 to 3, in, The situation parameter (S) gives whether there is speech in the environment, If speech is present, the dynamic scheme is modified in a manner that is beneficial to speech clarity; otherwise, the dynamic scheme is modified in a manner that is beneficial to sound quality.

5. The method according to any one of claims 1 to 3, in, modifying the dynamic scheme by changing at least one parameter of the dynamic scheme, The parameters are selected from a set of parameters including: start time, stop time (t_r), inflection point, compression ratio, and expansion ratio.

6. The method according to any one of claims 1 to 3, in, The situation parameter (S) is an environment category, so that the dynamic scheme is modified according to the environment category, In particular, the environment category is selected from a group consisting of: speech, music, and interference noise.

7. The method according to any one of claims 1 to 3, in, The situation parameter (S) is a movement pattern, so that the dynamic scheme is modified according to the movement of the hearing system (2), In particular, the movement mode is selected from a set of movement modes including: stationary position, walking movement, running movement, and driving.

8. The method according to any one of claims 1 to 3, in, The situation parameter (S) is the stability of the current acoustic environment, in particular the stability of the input signal, so that the dynamic scheme is modified according to the stability of the current acoustic environment.

9. The method according to any one of claims 1 to 3, in, The situation parameter (S) is the diffuseness of the current acoustic environment, so that the dynamic solution is modified according to the diffuseness of the current acoustic environment.

10. The method according to any one of claims 1 to 3, in, The situation parameter (S) is the distance between the hearing system (2) and a sound source in the current acoustic environment, in particular a speaker, so that the dynamic scheme is modified according to the distance.

11. The method according to any one of claims 1 to 3, in, The situation parameter (S) is the reverberation time of the current acoustic environment, so that the dynamic scheme is modified according to the reverberation time. The dynamic scheme has two time constants, namely the start time and the stop time. The reverberation time is repeatedly determined, and when the reverberation time increases, the at least one of the time constants is increased, whereas when the reverberation time decreases, the at least one of the time constants is decreased.

12. A method for operating a hearing system (2) in a current acoustic environment, in, The hearing system (2) has a signal processing device (6), The signal processing device (6) generates an output signal (A) from the input signal (E) by amplifying the input signal (E) with an amplification factor (V). The hearing system (2) has an earpiece (8), and the output signal (A) is output via the earpiece (8). wherein the amplification factor (V) is set by automatic gain adjustment with a dynamics processor (18), the dynamics processor (18) being capable of operating with a dynamic scheme defining the amplification factor (V) as a function of the level of the input signal (E), wherein the hearing system (2) is operated in a program, the program being associated with a dynamic scheme, with which the dynamic processor (18) is operated during operation in the program, wherein at least one situation parameter (S) is determined, the situation parameter characterizing the current acoustic environment, wherein, during the operation of the one program, the associated dynamic scheme is modified according to the situation parameter (S), so that the dynamic processor (18) is operated using the modified dynamic scheme, It is characterized in that Distinguish between the frequency range with speech (B2) and the frequency range without speech (B1, B3), By setting parameters for a frequency range (B1, B2, B3) depending on whether the corresponding frequency range (B1, B2, B3) is a speech frequency range (B2) or a frequency range (B1, B3) without speech, the parameters of the dynamic scheme are changed in a frequency-dependent manner, The situation parameter (S) is the signal-to-noise ratio of the current acoustic environment, For the speech frequency range (B2), in the case of a negative signal-to-noise ratio, at least one time constant is reduced, and in the case of a positive signal-to-noise ratio, the at least one time constant is increased, For a frequency range (B1, B3) without speech, the at least one time constant is increased when the signal-to-noise ratio is negative, and the at least one time constant is decreased when the signal-to-noise ratio is positive.

13. The method according to claim 12, in, The dynamic scheme is modified by changing the start time of the dynamic scheme or the stop time of the dynamic scheme (t_r) or both in a frequency-dependent manner.

14. The method according to claim 12, in, The dynamics processor (18) is a compressor, whereby the dynamics scheme is a compression scheme.

15. The method according to claim 12, in, The dynamics processor (18) is an expander, so that the dynamics scheme is an expansion scheme.

16. The method according to any one of claims 12 to 15, in, The situation parameter (S) gives whether there is speech in the environment, If speech is present, the dynamic scheme is modified in a manner that is beneficial to speech clarity; otherwise, the dynamic scheme is modified in a manner that is beneficial to sound quality.

17. The method according to any one of claims 12 to 15, in, modifying the dynamic scheme by changing at least one parameter of the dynamic scheme, The parameters are selected from a set of parameters including: start time, stop time (t_r), inflection point, compression ratio, and expansion ratio.

18. The method according to any one of claims 12 to 15, in, The situation parameter (S) is an environment category, so that the dynamic scheme is modified according to the environment category, In particular, the environment category is selected from a group consisting of: speech, music, and interference noise.

19. The method according to any one of claims 12 to 15, in, The situation parameter (S) is a movement pattern, so that the dynamic scheme is modified according to the movement of the hearing system (2), In particular, the movement mode is selected from a set of movement modes including: stationary position, walking movement, running movement, and driving.

20. The method according to any one of claims 12 to 15, in, The situation parameter (S) is the stability of the current acoustic environment, in particular the stability of the input signal, so that the dynamic scheme is modified according to the stability of the current acoustic environment.

21. The method according to any one of claims 12 to 15, in, The situation parameter (S) is the diffuseness of the current acoustic environment, so that the dynamic solution is modified according to the diffuseness of the current acoustic environment.

22. The method according to any one of claims 12 to 15, in, The situation parameter (S) is the distance between the hearing system (2) and a sound source in the current acoustic environment, in particular a speaker, so that the dynamic scheme is modified according to the distance.

23. The method according to any one of claims 12 to 15, in, The situation parameter (S) is the reverberation time of the current acoustic environment, so that the dynamic scheme is modified according to the reverberation time. The dynamic scheme has two time constants, namely the start time and the stop time. The reverberation time is repeatedly determined, and when the reverberation time increases, the at least one of the time constants is increased, whereas when the reverberation time decreases, the at least one of the time constants is decreased.

24. A hearing system (2), comprising: A signal processing device (6) which is designed to carry out the method according to any one of claims 1 to 23, The receiver (8) is used to output the output signal (A).

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