EP2222096A2 - Method for processing the signal of a hearing device and corresponding hearing device - Google Patents

Abstract

The method involves converting an input signal (e) into an intermediate signal (a11) according to conversion algorithm i.e. linear compression algorithm. The input signal is converted into another intermediate signal (a13) according to another conversion algorithm i.e. non-linear compression algorithm, parallel to conversion of the input signal according to the former conversion algorithm. An output signal (a) is formed from the intermediate signals with a mixing ratio that depends on the result of classification. An independent claim is also included for a hearing device comprising a conversion device.

Description

The present invention relates to a method for processing an input signal to an output signal in a hearing device by processing the input signal according to a first processing algorithm into a first intermediate signal and processing the input signal according to a second processing algorithm into a second intermediate signal in parallel with the processing of the input signal according to the first processing algorithm , Moreover, the present invention relates to a corresponding hearing device with a first processing device and a second processing device. The term "hearing device" is understood here to be any sound-emitting device that can be worn on the head or in or on the ear, in particular a hearing device, a headset, headphones and the like.

Hearing aids are portable hearing aids that are used to care for the hearing impaired. In order to meet the numerous individual needs, different types of hearing aids such as behind-the-ear hearing aids (BTE), hearing aid with external receiver (RIC: receiver in the canal) and in-the-ear hearing aids (IDO), e.g. Concha hearing aids or canal hearing aids (ITE, CIC). The hearing aids listed by way of example are worn on the outer ear or in the ear canal. In addition, bone conduction hearing aids, implantable or vibrotactile hearing aids are also available on the market. The stimulation of the damaged hearing takes place either mechanically or electrically.

Hearing aids have in principle as essential components an input transducer, an amplifier and an output transducer. The input transducer is usually a sound receiver, z. As a microphone, and / or an electromagnetic receiver, for. B. an induction coil. The output transducer is usually as an electroacoustic transducer, z. As miniature speaker, or as an electromechanical transducer, z. B. bone conduction, realized. The amplifier is usually integrated in a signal processing unit. This basic structure is in FIG. 1 shown using the example of a behind-the-ear hearing aid. In a hearing aid housing 1 for carrying behind the ear, one or more microphones 2 for receiving the sound from the environment are installed. A signal processing unit 3, which is also integrated in the hearing aid housing 1, processes the microphone signals and amplifies them. The output signal of the signal processing unit 3 is transmitted to a loudspeaker or earpiece 4, which outputs an acoustic signal. The sound is optionally transmitted via a sound tube, which is fixed with an earmold in the ear canal, to the eardrum of the device carrier. The power supply of the hearing device and in particular the signal processing unit 3 is effected by a likewise integrated into the hearing aid housing 1 battery. 5

A hearing aid wearer generally uses his hearing aid in different acoustic situations that place different demands on the signal processing in the hearing aid. So z. B. when listening to music a rather linear setting with little compression and regulation of the device to success, while for the understanding of language, especially in noisy environment, a rather non-linear setting, ie with compression, preferably with fast time constant benefits.

From the publication US 2007/0053535 A1 For example, a method for operating a hearing device is known. A signal from at least one signal source is recorded. At least one of these recorded signals is classified into one of several predefined sound classes. In doing so, properties of the sound source are considered. Finally, a hearing program is selected according to the classification result in the hearing aid.

Furthermore, the document discloses EP 1 829 028 A1 a method for adaptively adapting a sound processing parameter. The input signal is processed to achieve a specific dynamic range. In this case, a measured dynamic range is adapted to a desired dynamic range by the gain is set accordingly.

In addition, the document shows EP 1 307 072 A2 a hearing aid in which disturbing acoustic effects caused by switching operations should be avoided. The signal processing in the hearing device leads to sliding from a first operating state to a second operating state. During the switching process both operating states are simultaneously present in the hearing aid. The sliding transition occurs through parallel signal processing in at least two signal paths of the hearing aid, wherein a signal resulting from the first operating state and a signal resulting from the second operating state are added in alternating weighting.

Furthermore, in the document DE 10 2005 061 000 A1 a signal processing for hearing aids with multiple compression algorithms shown. For situation-related improvement of the signal processing, the input signal is classified with regard to the current hearing situation. Depending on the classification result, the input signal is amplified according to a first compression algorithm or a second compression algorithm. As a result, the respective advantages of the various compression algorithms in the individual listening situations can be exploited.

The object of the present invention is to be able to better adapt the signal processing of a hearing device to a situation.

According to the invention, this object is achieved by a method for processing an input signal to an output signal in a hearing device by processing the input signal according to a first processing algorithm to a first intermediate signal and processing the input signal according to a second processing algorithm to a second intermediate signal in parallel with processing the input signal according to the first processing algorithm, and classifying the input signal and forming the output signal from both the first and the second intermediate signal a mixing ratio that depends on the result of the classification.

In addition, the invention provides a hearing device with a first processing device for processing an input signal according to a first processing algorithm to a first intermediate signal, a second processing device for processing the input signal according to a second processing algorithm to a second intermediate signal in parallel with the processing of the input signal according to the first processing algorithm, further comprising classifying means for classifying the input signal, and third processing means for forming an output from each of the first and second intermediate signals with a mixing ratio which depends on the result of the classifying.

Advantageously, it is thus possible to mix different signal processing algorithms as desired in order better to be able to cope with a specific hearing situation. In particular, a mixing ratio of the output signals of the processing algorithms can be controlled by the classification result.

In the case of the first processing algorithm, a regulation preferably takes place with the level of the input signal and, in the case of the second processing algorithm, a regulation with the level of the second intermediate signal. When using the level of the input signal, for example, the compression rate, the gain or a time constant can be controlled. In the Using the level of the second intermediate signal, however, the frequency-dependent gain can be controlled.

In particular, the first or the second processing algorithm may each be a compression algorithm. In this case, it may be of particular advantage if the first processing algorithm is a linear compression algorithm (long time constant, eg 10 s) and the second processing algorithm is a non-linear compression algorithm (much shorter time constant eg 10 ms) at least in a given period of time , This allows the compression to be adapted very precisely to a specific situation.

In addition, the first processing algorithm may have a first time constant and the second processing algorithm may have a second time constant corresponding to the first time constant in the type, wherein the first time constant is different from the second time constant. As a result, depending on the situation, different time constants in the hearing device can be used.

Furthermore, the first processing algorithm may be based on a wideband and the second processing algorithm on a narrowband level measurement. Thus, both broadband signal processing and narrowband signal processing can simultaneously enter an output signal. In addition, the first processing algorithm may be input-related and the second output-related.

FIG 1

eine Prinzipskizze zum Aufbau eines Hörgeräts gemäß dem Stand der Technik;

FIG 2

ein Eingangssignal einer ersten oder zweiten Verarbeitungseinrichtung;

FIG 3

ein Ausgangssignal einer ersten Verarbeitungseinrichtung;

FIG 4

ein Ausgangssignal einer zweiten Verarbeitungseinrichtung;

FIG 5

ein weiteres Eingangssignal einer anderen ersten oder zweiten Verarbeitungseinrichtung;

FIG 6

ein Ausgangssignal der anderen ersten Verarbeitungseinrichtung;

FIG 7

ein Ausgangssignal der anderen zweiten Verarbeitungseinrichtung;

FIG 8

eine Hörgeräteverstärkung mit Eingangspegelregelung;

FIG 9

eine Hörgeräteverstärkung mit Ausgangspegelregelung und

FIG 10

einen schematischen Aufbau einer Hörvorrichtung gemäß einer Ausführungsform der vorliegenden Erfindung.

The present invention will be explained in more detail with reference to the accompanying drawings, in which:

FIG. 1

a schematic diagram of the construction of a hearing aid according to the prior art;

FIG. 2

an input signal of a first or second processing device;

FIG. 3

an output signal of a first processing device;

FIG. 4

an output of a second processing means;

FIG. 5

another input signal of another first or second processing means;

FIG. 6

an output of the other first processing means;

FIG. 7

an output of the other second processing means;

FIG. 8

a hearing aid amplification with input level control;

FIG. 9

a hearing aid amplification with output level control and

FIG. 10

a schematic structure of a hearing device according to an embodiment of the present invention.

The exemplary embodiments described in more detail below represent preferred embodiments of the present invention FIGS. 2 to 9 detailed in the principle of the invention.

In the case of multi-channel compression, for example, the presence of signal components of different intensities in the bands of the compression makes the amplification of the different signal components so unfortunate that the spectrum is smeared and thus the signal-to-noise ratio (SNR) is worsened. This can be, for example, based on the FIGS. 2 and 4 detect. FIG. 2 shows a useful signal n in a first channel k1 and a second channel k2 Interference signal s. The intensity I is plotted on the ordinate. The useful signal n and the interference signal s thus have the drawn signal-to-noise ratio SNR1. For non-linear amplification of the input signal from FIG. 2 can now the output signal from FIG. 4 arise. The amplified useful signal n _v has compared to the amplified interference signal s _v only the signal-to-noise ratio SNR2. SNR2 is smaller than SNR1 (at least output-related).

If, on the other hand, the determination of the intensity is broadband in another processing device, this results in accordance with FIG. 3 to the signals n ' _v and s' _v . This linear amplification preserves the SNR. Ie. SNR1 = SNR2 (output-related). It can thus be seen that the different processing devices (linear or non-linear amplification) can have different effects on a disturbed signal.

In addition, there is also a temporal effect, which can also result in a deterioration of the SNR. Here, rather, the time constants play a role. For example, according to FIG. 5 after a useful signal n (eg voice signal) an interference signal s (eg noise) with a lower intensity occur. FIG. 5 shows the respective levels L as a function of the time t. The signal-to-noise ratio is SNR1. Will now be appropriate FIG. 7 initially amplified by a very fast time constant, the useful portion to an amplified useful signal n _v , but then the energetically lower noise applied with a higher gain, whereby an amplified interference signal s _v is formed, the ratio SNR2 between useful and interference signal is degraded. This means: SNR1> SNR2. This negative effect is amplified by the use of fast time constants.

In contrast, in another processing device according to FIG. 6 used a slower time constant, so that after the amplification, the signals n ' _v and s' _v result, so can the signal-to-noise ratio SNR2 be kept constant, ie SNR1 = SNR2.

It would therefore be desirable if the compression characteristic, the time constants, the level meters used (narrowband or broadband, input or output related) would be automatically selected by the system as a function of the situation, thus automatically ensuring the hearing aid wearer the best compression characteristic in the respective acoustic situation ,

Basically, it is possible either to automatically optimize the compression (possibly to the detriment of the SNR) or to regulate a linear gain according to an AVC (Automatic Volume Control) with an output-side level control (the SNR is usually retained). The first alternative (automatic optimization of the compression parameters) can be performed according to FIG. 8 be implemented. A microphone 10 provides an input signal which is amplified by an amplifier 11. The level of the input signal (situation) is used to control the amplifier 11. The output signal of the amplifier 11 is forwarded to a handset 12. In addition to a gain but also the compression rate or a time constant can be controlled using the input signal. In this way, for example, a hearing program could be switched depending on the situation so that the compression parameters (amplification, compression) are adapted to the respective situation. In this case, in a pre-stage of the invention, only a switching between different compression settings would take place, which were previously created during a fine adjustment together with the hearing device wearer. However, this system does not change the level meter used to adjust the gain, so it does not toggle between input-related and output-related compression.

Other systems with input-side control could allow a situational selection of time constants.

In this case, the time constants in the respective channel could be determined adaptively, in particular as a function of the (narrowband) level. This counteracts deterioration of SNR in the time domain, but spectral blurring remains a problem.

When approaching the whole problem can also be in FIG. 9 be considered alternative presented. There, the supplied from the microphone 10 input signal is also supplied to an amplifier 11, the output signal is passed to the handset 12. However, here takes place a feedback of the output signal of the amplifier 11 and thus in particular an output side, slow level control. The system is similar to an AVC with the difference that the resulting frequency-specific gain is determined from a complex level statistic in several bands (eg 128). Not only purely physical but also psychoacoustic factors can be taken into account (see the above-mentioned EP 1 829 028 A1 ). In addition, because the system regulates very slowly and therefore operates linearly within the time constant (several seconds), it allows a pleasant sound and a pleasant loudness perception in a wide variety of acoustic environments. The disadvantage of this system is that, especially in those cases in which the hearing-impaired person only has a very narrow residual dynamic (frequency-dependent difference between discomfort threshold UCL and hearing threshold HS) (eg <30 dB), the processed signal is not complete can be mapped to the dynamic range. This has the consequence that the understanding of speech, especially in acoustic situations with background noise can only be improved insufficiently.

As the examples above show, the benefit of each system depends on the acoustic situation. It is therefore according to the invention in FIG. 10 sketched system provided. The input signal, ie the signal e generated by the microphone 10, is in a first branch of a first Processing means 11, which is controlled with the input signal supplied. A corresponding output signal a ₁₁ is made available. In a second branch, the input signal e from the microphone 10 is supplied to a second processing device 13, which has an output level control here. There, an output signal a _{13 is} generated. The input signal e of the microphone 10 is finally fed to a classifier 14 in a third branch. The classification result 14 is used in a weighting unit 15 to generate weights g ₁₁ and g ₁₃ corresponding to the output signals a ₁₁ and a ₁₃ . In the weighting unit 15, the two output signals a ₁₁ and a _{13 are combined} with the respective weights g ₁₁ and g ₁₃ so that a mixed output signal a results at the output of the weighting unit 15, which is supplied to the handset 12. In the first branch, for example, depending on the situation, the compression rate, the gain or a time constant can be regulated. In contrast, in the second branch, for example, the frequency-dependent amplification can be regulated. It is thus possible to achieve a continuous mixing of two parallel output signals a ₁₁ , a ₁₃ during operation, the mixing ratio depending on the classification result.

If the algorithm underlying the first processing device 11 is an AGCi (Automatic Gain Control Input Dependent) and the algorithm underlying the second processing device 13 is an AGCo (Automatic Gain Control Output Dependent), for example, the gain may be 70% out of the value in a special situation AGCo and 30% from the value of AGCi. Thus, for example, a hard switching between one of the two systems can be avoided and achieve a continuous mixture. Similarly, mixed signals from quasi-linear and non-linear compression systems, processing devices with different time constants and / or processing devices with evaluation of either a wideband or several narrowband Level meter can be realized. The mixing ratio is determined by the classification system and the classifier 14, respectively.

By combining different systems (first processing device 11 and second processing device 13), on the one hand, the SNR that is important for speech understanding can be optimized in situations in which speech understanding plays a role. In contrast, in situations in which a pleasant sense of loudness plays the decisive role, for. B. in a noisy environment to optimize the listening effort, the system can switch to a more linear system, which simultaneously sets the basic gain so that the output of the hearing aid is perceived by the individual hearing aid wearer as pleasant. If the hearing aid wearer is in a situation in which the useful signal and noise lie in different channels, the system can automatically switch completely or partially to the evaluation of the broadband power meter so that there is no different gain in the different channels and thus keep the SNR constant can be.

Claims (10)

A method of processing an input signal to an output signal in a hearing device - Processing the input signal (e) according to a first processing algorithm to a first intermediate signal (a ₁₁ ) and Processing the input signal (e) according to a second processing algorithm into a second intermediate signal (a ₁₃ ) in parallel with the processing of the input signal (e) according to the first processing algorithm,
marked by Classifying the input signal (e) and - Forming the output signal (a) from both the first and the second intermediate signal (a ₁₁ , a ₁₃ ) with a mixing ratio, which depends on the result of the classification. The method of claim 1, wherein in the first processing algorithm, a control with the level of the input signal (e) and in the second processing algorithm, a control with the level of the second intermediate signal (a ₁₃ ). The method of claim 1 or 2, wherein each of the first and second processing algorithms is a compression algorithm. The method of claim 3, wherein the first processing algorithm is a linear compression algorithm at least in a predetermined time period and the second processing algorithm is a non-linear compression algorithm. Method according to one of the preceding claims, wherein the first processing algorithm has a first time constant and the second processing algorithm has a second time constant corresponding to the first time constant in the type, and wherein the first time constant is different from the second time constant. Method according to one of the preceding claims, wherein the first processing algorithm based on a broadband and the second processing algorithm based on a narrow-band level measurement. Hearing device with a first processing device (11) for processing an input signal (e) according to a first processing algorithm into a first intermediate signal (a ₁₁ ), second processing means (13) for processing the input signal (e) according to a second processing algorithm into a second intermediate signal (a ₁₃ ) in parallel with the processing of the input signal (e) according to the first processing algorithm,
marked by a classification device (14) for classifying the input signal (e) and - Third processing means (15) for forming an output signal (a) from both the first and the second intermediate signal (a ₁₁ , a ₁₃ ) with a mixing ratio, which depends on the result of the classifying. Hearing apparatus according to claim 7, wherein in the first processing means (11) the level of the input signal (e) and in the second processing means (13) the level of the second intermediate signal (a ₁₃ ) is used for control. Hearing apparatus according to claim 7 or 8, wherein the first and the second processing algorithm is each a compression algorithm. Hearing apparatus according to one of claims 7 to 9, wherein the first processing means (11) comprises a level meter for broadband level measurement and the second processing means (13) a level meter for narrow-band level measurement having.

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