METHOD FOR OPERATING A BINAURAL HEARING DEVICE, BINAURAL HEARING DEVICE AND COMPUTER PROGRAM

Description

[0001] The invention relates to a method for operating a binaural hearing aid with two individual devices, each of which comprises at least one input converter for receiving an acoustic signal and converting it into a multi-channel input signal, an impulse noise suppression unit for generating an attenuation curve to reduce impulse noise signal levels in the input signals, a signal processing unit for multi-channel signal amplification of the input signals and generation of an output signal, an output converter for converting the output signal into an audio signal, and a transmitter and receiver unit for signal coupling between the individual devices. The invention further relates to a binaural hearing aid and software on a data carrier for carrying out the method.

[0002] Hearing aids are portable devices designed to provide hearing assistance to people with hearing loss or impairment. To meet a wide range of individual needs, various hearing aid designs are available, including behind-the-ear (BTE) hearing aids, receiver-in-the-canal (RIC) hearing aids, and in-the-ear (ITE) hearing aids, such as concha or in-the-canal (ITE) hearing aids. The hearing aids listed above are worn on the outer ear or in the ear canal. In addition, bone conduction hearing aids, implantable hearing aids, and vibrotactile hearing aids are also available. These devices stimulate the impaired hearing either mechanically or electrically.

[0003] Such hearing aids essentially consist of an input transducer, an amplifier, and an output transducer as their main components. The input transducer is usually an acousto-electrical transducer, such as a microphone, and / or an electromagnetic receiver, for example, an induction coil or a (radio frequency, RF) antenna. The output transducer is usually an electro-acoustic transducer, for example, a miniature loudspeaker (receiver), or an electromechanical transducer, such as a bone conduction receiver. The amplifier is typically integrated into a signal processing unit. Power is usually supplied by a battery or a rechargeable battery.

[0004] The input signals received by the input converters are typically multi-channel, meaning that the input signals are divided into several individual frequency channels, with each frequency channel covering a frequency band of a certain spectral width. For example, a hearing aid might have 48 (frequency) channels in a frequency range between 0 kHz (kilohertz) and 24 kHz, where the individual signal components of the input signal in the channels can be processed individually by the signal processing unit, in particular individually filtered, amplified and / or attenuated.

[0005] In a binaural hearing aid, the user wears two such devices on opposite sides of the head, with each device assigned to one ear. A communication link exists between the devices. During operation, data, potentially large amounts of data, is wirelessly exchanged between the hearing aids in the right and left ears. The exchanged data and information enable particularly effective adaptation of the individual devices to the specific acoustic environment. This results in a particularly authentic spatial sound for the user and improves speech intelligibility, even in noisy environments.

[0006] During operation, the hearing aid settings, i.e., one or more hearing aid parameters and / or hearing aid performance, are automatically adjusted based on various parameters or sizes, so that the most suitable hearing signal is generated for the user in every (acoustic) environment or listening situation.

[0007] The sudden occurrence of impulse noise can negatively affect the automatic adjustment of hearing aid parameters and / or hearing aid performance. Impulse noise, in this context, refers specifically to an acoustic sound event with a very rapid or sudden rise time of the signal level (less than 0.2 s), which exhibits large signal components at certain frequencies. Impulse noise occurs particularly with loud, popping sounds, such as clapping hands, clattering dishes, or a slamming door. Special algorithms, circuits, and programming techniques are used to attenuate such impulse noise signals without reducing the speech quality of the original signal.

[0008] The measures used to dampen or suppress impulse noise in hearing aids are referred to here and in the following as impulse noise suppression or impulse noise filtering. With such impulse noise suppression, a detection process for impulse noise recognition runs continuously in the background, so that when a noise with a very rapidly increasing amplitude (impulse noise) is detected, the amplification of corresponding frequencies can be suppressed as instantaneously as possible to ensure optimal listening comfort. Impulse noise suppression generates an attenuation value for each channel of the input signal, which indicates how much the signal amplification of the signal processing unit in each channel should be dampened or reduced to minimize the disruptive effects of the impulse noise in the output signal.The attenuation values ​​for all frequency channels are also collectively referred to as an attenuation curve.

[0009] In binaural hearing aids, the impulse noise suppression of the individual devices is usually not synchronized with each other, because it is typically not possible in the short time during the impulse noise to adjust the amplification in the left and right individual devices for several frequency bands or channels to each other, because the transmission of, for example, 48 amplification or attenuation values ​​would take too long.

[0010] In practice, the signal levels of impulse noise detected by individual hearing aids can differ due to ambient reverberation and the acoustic influence of the wearer's head (head shadowing). As a result, the impulse noise suppression of the left and right hearing aids produces different attenuation curves and thus different amplifications for the output signals. This can lead to fluctuations or deviations in the natural interaural level differences (ILDs). This effect can impair the hearing aid user's localization and spatial perception of the acoustic environment.For example, it may happen that the level of the impulse signal in the output signal is reduced more for the individual device that is spatially closer to the impulse sound signal source than for the more distant individual device, which causes a brief change in the user's perception of direction.

[0011] From EP 3 337 186 A1, a hearing aid and a method for attenuating impulse sound signals are known. For this purpose, a recorded audio signal is processed into a processed audio signal in response to an impulse sound. The processing can include reducing the amplification of the output sound signal when an impulse sound is detected.

[0012] The invention is based on the objective of providing a particularly suitable method for operating a binaural hearing aid. In particular, changes in the spatial directional perception of the acoustic environment when impulse sound occurs are to be reduced. The invention is further based on the objective of providing a particularly suitable binaural hearing aid and particularly suitable software on a data carrier.

[0013] With regard to the method, the problem is solved according to the invention by the features of claim 1, with regard to the binaural hearing aid by the features of claim 9, and with regard to the software by the features of claim 10. Advantageous embodiments and further developments are the subject of the dependent claims.

[0014] The advantages and features mentioned with regard to the procedure can also be applied analogously to the hearing aid and / or the software, and vice versa.

[0015] If the following process steps are described, advantageous designs for the hearing aid result in particular from the fact that it is designed to perform one or more of these process steps.

[0016] The method according to the invention is designed and configured for operating a binaural hearing aid. The hearing aid is binaural and comprises two individual devices, each of which includes at least one input transducer, impulse noise suppression, a signal processing unit, a transmitter and receiver, and at least one output transducer. These devices are configured to receive sound signals from the environment and output them to the user of the hearing aid. The aforementioned device components are housed in separate hearing aid casings. The casings are designed so that they can be worn by the user on the head and near the ear, e.g., in the ear, on the ear, or behind the ear. For example, the hearing aid is designed as a back-to-the-ear (BTE) hearing aid, an in-the-ear (ITO) hearing aid, or a receiver-in-canal (RIC) hearing aid.

[0017] With a binaural hearing aid, the user wears the two individual devices on opposite sides of the head, so that each device is assigned to one ear. The individual devices are equipped for signal data exchange via a wireless interface formed by the transmitting and receiving units (transceivers).

[0018] The hearing aid is primarily intended for use by a hearing-impaired user. The hearing aid is designed to receive sound signals from the environment and output them to the user. For this purpose, the hearing aid has at least one input transducer, in particular an acousto-electrical transducer, such as a microphone. During operation, the input transducer receives sound signals (noises, tones, speech, etc.) from the environment and converts each of these into an electrical input signal. This input signal is multi-channel. In other words, the acoustic signals are converted into a multi-channel input signal. The input signal therefore has several frequency channels, in particular at least two, preferably at least 20, and especially preferably at least 40, for example 48, each covering an assigned frequency band within the hearing aid's frequency range.For example, a frequency range between 0 kHz and 24 kHz is divided into 48 channels, so that input signals with 48 channels are generated.

[0019] Each individual device has an output converter, in particular an electro-acoustic converter, such as a headphone. An electrical (multi-channel) output signal is generated from the electrical (multi-channel) input signal by modifying the input signal, or the individual frequency or signal channels, in a signal processing unit (e.g., amplifying, filtering, attenuating).

[0020] To dampen or suppress impulse noise, each individual device incorporates impulse noise suppression (impulse noise filtering), which is integrated, for example, into the signal processing unit. Preferably, the impulse noise suppression is placed upstream of a signal amplifier in the signal processing unit.

[0021] Impulse noise suppression is designed and configured to detect the occurrence of impulse noise in the channels of the input signal and to generate an attenuation value for each channel, reducing the amplifier's signal gain for that channel by the amount of attenuation. The occurrence of impulse noise can be detected, for example, by a very rapid (< 0.2 s) increase in signal amplitude (signal level). The set of attenuation values ​​for several or all channels of the input signal is subsequently referred to as the attenuation curve. If no impulse noise is detected, the attenuation values ​​will, for example, be 0 dB (decibels), so that the signal gain in the signal processing unit or amplifier is not affected.In the case of impulse noise, the attenuation values ​​can, for example, range from -20 dB to -40 dB, whereby the subsequent signal amplification is attenuated or reduced by this amount. The attenuation curves generated by the impulse noise suppression during the operation of the hearing aid thus reduce the impulse noise signal levels in the amplified input and output signals.

[0022] The procedure involves determining a scalar limit value from the respective attenuation curve in each individual device. In other words, a single scalar limit value is determined from the multiple attenuation values, for example, 48. This limit value can be determined from the complete attenuation curve for all channels or only from a (partial) attenuation curve of the channels affected by the impulse noise. For example, the impulse noise of a slamming door has higher signal components in the low-frequency bands or channels, while the clatter of dishes has more signal components in higher-frequency channels.

[0023] The limit values ​​of the individual devices are then transmitted to the other individual device, whereby a common, (pseudo-)synchronized limit value is determined from the limit values ​​of both devices. This common or synchronized limit value is then used to limit the attenuation curves. In other words, the synchronized limit value is used, in particular, as a lower threshold, whereby if the attenuation values ​​or the attenuation curve reach or fall below the limit value, they are restricted or limited to the limit value. The limited attenuation curve or the limited attenuation values ​​are used to adjust the signal gains in the signal processing unit.

[0024] This method accepts the possibility that, due to the synchronized limit value, one of the individual devices might not sufficiently attenuate the impulse sound in the output signal. However, this ensures that the user's directional perception is not adversely affected by impulse sounds. This improves localization and spatial perception of the acoustic environment. In particular, this prevents fluctuations or deviations in the natural interaural level differences (IDL), resulting in a particularly suitable method for operating a binaural hearing aid.

[0025] The method according to the invention aims not to synchronize the exact gains in the frequency bands (channels), but rather to synchronize a scalar (broadband) limit value that specifies a maximum attenuation value for all channels. This limits the previously essentially unlimited attenuation values ​​of a relevant number of frequency bands (the bands with the highest desired attenuation) to the same limit value when this value is reached, exceeded, or fallen below.

[0026] The problem of very fast synchronization (microsecond range) of multi-channel amplification on both sides of the ear is thus solved according to the invention by a (pseudo-)synchronization in which a single limit value is transmitted between the individual devices. The limit values ​​essentially correspond to an expected (estimated, extrapolated) value for the next impulse sound (impulse sound event), whereby the limit values ​​between impulse sounds are transmitted and synchronized between the individual devices. The synchronized limit value applies essentially only to the next impulse sound. The method is based on the assumption that a subsequent (future) impulse sound has a comparable amplitude or signal level to the current (past) impulse sound, as is the case, for example, with hand clapping or the rattling of dishes.If this assumption is true, the applied attenuation values ​​in the relevant frequency ranges (those with the most desired attenuation) are limited to the synchronized maximum attenuation value (limiting value) in both individual devices and are therefore equal and flat. This uniformity or flatness of the attenuation curve (same attenuation value for different frequencies) is a further advantage of the method: it prevents the impulse sound from being distorted in its frequency shape, at least across the frequency range and during the time in which the synchronized limiting value is applied.

[0027] Preferably, only one scalar limit value is exchanged between the individual devices. However, it is also conceivable that the frequency channels of the individual devices are divided into at least two frequency bands or frequency sections, and that a scalar limit value is determined and exchanged for each frequency band or section. For example, the frequency channels are divided into high frequencies and low frequencies, with low frequencies covering, for example, the frequency range of a slamming door and high frequencies covering, for example, the frequency range of rattling dishes. It is essential that a significantly reduced number of scalar limit values ​​is exchanged between the individual devices compared to the number of frequency channels, thus enabling rapid (pseudo-)synchronization of the individual devices.The individual devices transmit, for example, fewer than five scalar limit values ​​each, in particular fewer than three scalar limit values, preferably only one scalar limit value.

[0028] If no impulse sound occurs for an extended period, the probability that the next impulse sound is related to the previous one decreases. In a suitable implementation, the common or synchronized limit value is reset to a stored default value after a predetermined time. For example, as the time between impulse sounds increases, the synchronized limit value gradually or successively assumes a stored or predetermined default value, such as -20 dB. After a predetermined time period, at which the probability that the next impulse sound is related to the previous one is sufficiently low, the synchronized limit value is thus forgotten (the "forgetfulness factor"). What time period or probability is considered sufficiently low, and how large that probability actually is, is initially irrelevant.This can be determined, for example, from past data or from relevant trials or tests. Different time durations may result for different impulse sound events (door slamming, hand clapping, etc.), environmental / listening situations, or application scenarios.

[0029] In a possible further development approach, the common limit value is determined by averaging the two limit values. In other words, the limit values ​​are averaged to obtain the common or synchronized limit value. This results in a particularly simple and cost-effective determination of the common limit value.

[0030] In a preferred embodiment, a mean value and a minimum value of the corresponding damping curve are determined for each limit value. In other words, the damping values ​​determined by the impulse noise suppression are averaged to obtain a (damping) mean value, and a minimum damping value is determined as the (damping) minimum value. The limit value is suitably determined such that it lies between the mean value and the minimum value. The mean value thus represents an upper limit or threshold, and the minimum value a lower limit or threshold for the limit value. The determined limit value is therefore always greater than the minimum value and always less than the mean value.

[0031] In a suitable implementation, the limit value is determined from the sum of the mean and minimum values, where the mean and minimum values ​​are preferably modified by a weighting factor. The weighting factor conveniently has a range between zero (0) and one (1). This means that the weighting factor is greater than or equal to zero (≥ 0) and less than or equal to one (≤ 1). For example, the minimum value (min) is multiplied by the weighting factor (w), the mean value (mean) is multiplied by a factor of one minus the weighting factor (1 - w), and the weighted values ​​are added together. Expressed as a formula, the limit value (att_bb_thr) is thus, for example, as: att _ bb _ thr = w * min + 1 − w * mean .

[0032] The weighting factor can be a stored or predefined value. Preferably, the weighting factor is constant and dimensioned between 0 and 1 according to the specific requirements, depending on whether less (average, w = 0) or more (minimum, w = 1) attenuation is desired on average. The specific value of the weighting factor is initially of secondary importance. A suitable weighting factor can be determined, for example, from past data or from relevant tests or trials. Different weighting factors may be required for different impulse sound events (door slamming, hand clapping, etc.), environmental / listening situations, or application scenarios. The weighting factor can therefore be set, for example, depending on the current listening situation.

[0033] The binaural hearing aid according to the invention comprises two individual devices. Each individual device includes at least one input converter for receiving an acoustic signal and converting it into a multi-channel input signal, an impulse noise suppressor for generating an attenuation curve to reduce impulse noise signal levels in the input signal, a signal amplifier for multi-channel signal amplification of the input signal and generation of an output signal, an output converter for converting the output signal into an audio signal, and a transmitter and receiver unit for signal coupling between the individual devices. The impulse noise suppressor and the amplifier are, for example, part of a signal processing unit.

[0034] The hearing aid, in particular the signal processing unit or the impulse noise suppression unit, further comprises a controller, i.e., a control unit. The controller is generally configured – programmatically and / or circuit-wise – to carry out the method according to the invention described above. Specifically, the controller is thus configured to determine a limit value from the attenuation values ​​or attenuation curves and transmit it to the transmitting and receiving unit, as well as to determine a common or synchronized limit value from the limit value and the limit value transmitted by the other individual device, and to apply this to the attenuation curve as a limit or restriction.

[0035] In a preferred embodiment, the controller is formed, at least in its core, by a microcontroller comprising a processor and a data memory. The functionality for carrying out the inventive method is implemented programmatically in the form of operating software (firmware), so that the method—optionally in interaction with a hearing aid user—is carried out automatically when the operating software is executed in the microcontroller. Alternatively, within the scope of the invention, the controller can also be formed by a non-programmable electronic component, such as an application-specific integrated circuit (ASIC), in which the functionality for carrying out the inventive method is implemented by circuit design.

[0036] An additional aspect of the invention provides for software on a medium or data carrier for carrying out the method described above. This means that the software is stored on a data carrier and is designed and configured for carrying out the method described above. This results in particularly suitable software for operating a binaural hearing aid, with which the functionality for carrying out the method according to the invention is implemented programmatically. The software is therefore, in particular, operating software (firmware), with the data carrier being, for example, a data storage device of the controller.

[0037] An embodiment of the invention is explained in more detail below. It shows, in schematic and simplified representations: Fig. 1 a binaural hearing aid with two individual devices, and Fig. 2 a block diagram for the functional division of a method for operating the hearing aid.

[0038] Corresponding parts and sizes are always marked with the same reference symbols in all figures.

[0039] The Fig. 1 Figure 1 shows the basic structure of a binaural hearing aid 2 according to the invention. The hearing aid 2 is designed with two signal-linked hearing aids or individual devices 4a, 4b. The individual devices 4a, 4b are designed as behind-the-ear (BTE) hearing aids by way of example. The individual devices 4a, 4b are or can be linked to each other by means of a wireless communication link 6.

[0040] The communication link 6 is, for example, an inductive coupling between the individual devices 4a and 4b; alternatively, the communication link 6 can be implemented, for example, as a radio link, in particular as a Bluetooth or RFID link, between the individual devices 4a and 4b.

[0041] In the application state, the individual device 4a is, for example, positioned on the right ear of the hearing aid user, while the individual device 4b is positioned on the left ear.

[0042] The structure of the individual devices 4a and 4b is explained below using device 4a as an example, whereby the explanations can also be applied analogously to device 4b. The components of device 4a are identified by the suffix "a", while the corresponding components of device 4b are identified by the corresponding suffix "b" in the figures.

[0043] The individual device 4a comprises, as in the Fig. 1 A device housing 8a, shown schematically, contains one or more microphones, also referred to as (acousto-electrical) input transducers 10a. The input transducers 10a capture sound or acoustic signals in the vicinity of the hearing aid 2 and convert them into electrical, multi-channel input signals 12a. Fig. 2 Preferably, the input signals 12a have several frequency channels, for example 48 channels in the frequency range between 0 kHz and 28 kHz.

[0044] A signal processing unit 14a, which is also integrated into the device housing 8a, processes the input signals 12a. An output signal 16a ( Fig. 2 The signal from the signal processing unit 14a is transmitted to an output transducer 18a, which may be a loudspeaker or receiver, for example, and which outputs an acoustic signal. In the case of the individual device 4a, the acoustic signal may be transmitted to the eardrum of a hearing aid user via a sound tube (not shown) or an external receiver with an earmold that sits in the ear canal. However, an electromechanical output transducer 20, such as in a bone conduction receiver, is also conceivable as the receiver.

[0045] The power supply for the individual device 4a and in particular for the signal processing unit 14a is provided by a battery 20a which is also integrated into the device housing 8a.

[0046] The signal processing unit 14a is connected to a transmit and receive unit (transceiver) 22a. The transceiver 22a serves in particular for transmitting and receiving wireless signals via the communication link 6.

[0047] The signal processing unit 14a comprises an impulse noise suppressor 24a and a (signal) amplification or amplifier 26a, as well as a controller (not shown in detail) as a control unit. The controller is provided for, and is suitable and configured for, carrying out a method according to the invention for operating the hearing aid 2. The method is described below with reference to the Fig. 2 explained in more detail.

[0048] During operation of the hearing aid, the input transducers 10a, 10b receive two sound signals (noises, tones, speech, etc.) from the environment and convert them into the multi-channel input signals 12a, 12b. The input signals 12a, 12b are fed to the respective impulse noise suppressors 24a, 24b, which analyze the input signals 12a, 12b for the presence of impulse noise. If impulse noise is detected, the impulse noise suppressor 24a, 24b generates an attenuation curve 28a, 28b for controlling and / or regulating the multi-channel amplifier 26a, 26b.

[0049] The damping curves 28a, 28b are in the Fig. 2 This is shown schematically and exemplarily using a frequency-attenuation diagram, where the frequency f, for example from 0 kHz to 24 kHz, is plotted horizontally along the abscissa (x-axis), and the attenuation (gain), for example from -40 dB to 0 dB, is plotted along the vertical ordinate (y-axis). Attenuation curves 28a and 28b show, for example, the behavior of a slamming door whose impulse sound is low-pitched or low-frequency, so that the highest signal level—and correspondingly the lowest attenuation values—occur at low frequencies. Due to reverberation or head shadowing, attenuation curves 28a and 28b exhibit different shapes.

[0050] The controller of each individual device 4a, 4b determines a minimum value 30a, 30b and a mean value 32a, 32b from the respective damping curve 28a, 28b. The minimum value 30a, 30b corresponds to the lowest damping value of the damping curve 28a, 28b, i.e., the value with the greatest damping for the gain 26a, 26b, where the mean value 32a, 32b is the averaged value of the respective damping curve 28a, 28b.

[0051] The controller of each individual device 4a, 4b determines a scalar limit value 34a, 34b from the minimum value 30a, 30b and the mean value 32a, 32b, which lies between the mean value 32a, 32b and the minimum value 30a, 30b. The limit values ​​34a, 34b are calculated in particular according to the following formula: att _ bb _ thr = w * min + 1 − w * mean , where att_bb_thr is the limiting value 34a, 34b, min is the minimum value 30a, 30b, mean is the mean value 32a, 32b, and w is a weighting factor between zero and one (0 ≤ w ≤ 1).

[0052] The limit values ​​34a and 34b of the individual devices 4a and 4b are then transmitted to the other individual device 4b and 4a via the communication link 6. Thus, both limit values ​​34a and 34b are available to the controllers of both individual devices 4a and 4b. The controllers determine a synchronized limit value 36 from the limit values ​​34a and 34b by averaging them, and this synchronized limit value has the same value in both individual devices 4a and 4b. The limit value 36 is, for example, the arithmetic mean of the limit values ​​34a and 34b.

[0053] The values ​​for the scalar quantities of the minimum value 30a, 30b, the mean value 32a, 32b and the limiting values ​​34a, 34b, 36 are shown in the diagrams of the Fig. 2 schematically represented as dashed lines.

[0054] The synchronized limit value 36 is then used to limit the attenuation curves 28a and 28b. In other words, the synchronized limit value 36 is used specifically as a lower threshold; all values ​​of the respective attenuation curves 28a and 28b that reach or fall below this limit value 36 are limited to the limit value 36. In other words, the attenuation curves 28a and 28b are "clipped" below the limit value 36, so that the limited attenuation curves 28a' and 28b' are flattened in the region of highest spectral energy, thus preserving the spectral characteristics of the impulse sound during amplification. The limited attenuation curves 28a and 28b are used to adjust the signal gains of the (frequency) channels in the amplifier 26a and 26b.

[0055] The synchronized limit value 36 applies essentially only to the next impulse sound. The procedure is based on the assumption that a subsequent (future) impulse sound will have a comparable amplitude or signal level to the current (past) impulse sound. If this assumption is met, the attenuation curves 28a and 28b for the future impulse sound in both individual devices 4a and 4b are limited to the limit value 36, so that the attenuation curves 28a' and 28b' are essentially identical and flat. The synchronized limit value 36 is reset to a stored default value after a predetermined period. Reference symbol list

[0056] 2 Hearing aid 4a, 4b Individual device 6 Communication link 8a, 8b Device housing 10a, 10b Input converter 12a, 12b Input signal 14a, 14b Signal processing unit 16a, 16b Output signal 18a, 18b Output converter 20a, 20b Battery 22a, 22b Transceiver unit 24a, 24b Impulse noise suppression 26a, 26b Amplifier 28a, 28b, 28a', 28b' Attenuation curve 30a, 30b Minimum value 32a, 32b Average value 34a, 34b Limit value 36 Limit value

Claims

1. Method for operating a binaural hearing aid (2) having two individual devices (4a, 4b), wherein the individual devices (4a, 4b) each comprise a) at least one input transducer (10a, 10b) for recording an acoustic signal and converting it into a multichannel input signal (12a, 12b), wherein the multichannel input signal (12a, 12b) has at least two frequency channels, which each cover an assigned frequency band of a frequency range of the hearing aid (2) b) a pulsed sound suppression unit (24a, 24b) for recognizing an occurrence of pulsed sound in the frequency channels of the input signal (12a, 12b), and for generating a damping curve (28a, 28b) for reducing pulsed sound signal levels (12a, 12b) if an occurrence of pulsed sound is recognized, wherein the damping curve comprises a damping value in each case for each frequency channel of the input signal, c) an amplifier (26a, 26b) for multichannel signal amplification of the input signal (12a, 12b) and generating an output signal (16a, 16b) on the basis of the damping curve (28a, 28b), d) an output transducer (18a, 18b) for converting the output signal (16a, 16b) into a sound signal, and e) a transmitting and receiving unit (22a, 22b) for signal coupling between the individual devices (4a, 4b), - wherein in each individual device (4a, 4b), a scalar limiting value (34a, 34b) is determined from the respective damping curve (28a, 28b) characterized in that - the scalar limiting values (34a, 34b) are transferred to the respective other individual device (4a, 4b), - a common limiting value (36) is determined from the two limiting values (34a, 34b), - the damping curves (28a, 28b) are limited using the common limiting value (36), and - the signal amplification is set on the basis of the limited damping curves (28a', 28b').

2. Method according to Claim 1, characterized in that the common limiting value (36) is set after a predetermined period of time to a stored standard value if no further pulsed sound is recognized during the period of time.

3. Method according to Claim 1 or 2, characterized in that the common limiting value (36) is determined by averaging the two scalar limiting values (34a, 34b).

4. Method according to any one of Claims 1 to 3, characterized in that for the determination of the respective scalar limiting value (34a, 34b) from the damping curve (28a, 28b), a mean value (32a, 32b) and a minimum value (30a, 30b) of the damping curve (28a, 28b) are determined.

5. Method according to Claim 4, characterized in that the scalar limiting value (34a, 34b) is determined such that it is between the mean value (32a, 32b) and the minimum value (30a, 30b).

6. Method according to Claim 4 or 5, characterized in that the limiting value (34a, 34b) is determined by summing the minimum value (30a, 30b) and the mean value (32a, 32b).

7. Method according to any one of Claims 4 to 6, characterized in that the mean value (32a, 32b) and the minimum value (30a, 30b) are modified using a weighting factor to determine the scalar limiting value (34a, 34b).

8. Method according to Claim 7, characterized in that the weighting factor is dimensioned between zero and one.

9. Binaural hearing device (2) having two individual devices (4a, 4b), wherein the individual devices (4a, 4b) each comprise - at least one input transducer (10a, 10b) for recording an acoustic signal and converting it into a multichannel input signal (12a, 12b), wherein the multichannel input signal (12a, 12b) has at least two frequency channels, which each cover an assigned frequency band of a frequency range of the hearing aid (2), - a pulsed sound suppression unit (24a, 24b) for recognizing an occurrence of pulsed sound in the frequency channels of the input signal (12a, 12b), and for generating a damping curve (28a, 28b) for reducing pulsed sound signal levels in the input signal (12a, 12b) when an occurrence of pulsed sound is recognized, wherein the damping curve has a damping value in each case for each frequency channel of the input signal, - an amplifier (26a, 26b) for multichannel signal application of the input signal (12a, 12b) and for generating an output signal (16a, 16b) on the basis of the damping curve (28a, 28b), - an output transducer (18a, 18b) for converting the output signal (16a, 16b) into a sound signal, - a transmitting and receiving unit (22a, 22b) first signal coupling between the individual devices (4a, 4b), and - a controller for carrying out a method according to any one of Claims 1 to 8.

10. Software on a data carrier for carrying out a method according to any one of Claims 1 to 8 when the software runs on one controller of each of the two individual devices (4a, 4b).