EP4247007A1 - Method for operating a binaural hearing system - Google Patents

Abstract

The invention describes a method for operating a binaural hearing system (10) with a first hearing instrument (11) and a second hearing instrument (12), wherein ambient sound (18) is input through an electroacoustic first input transducer (M1) of the first hearing instrument (11). first input signal (E1) is generated, and a second input signal (E2) is generated by an electroacoustic second input transducer (M2) of the second hearing instrument (12), a first instantaneous amplification parameter (G1) being determined based on the first input signal (E1), and a second instantaneous gain parameter (G2) is determined based on the second input signal (E2), wherein a first parameter (P1) of an automatic gain control for the first input signal (E1) and / or a second parameter (P2) of an automatic gain control for the second Input signal (E2) is adjusted in such a way that, as a result of said adjustment (41, 42), a difference between the first and second instantaneous amplification parameters (G1, G2) is reduced, and in the first and second hearing instruments (11, 12), respectively Signal processing of the first or second input signal (E1, E2) takes place with the first or second parameter (P1, P2) of the automatic gain control adapted in this way.

Description

The invention relates to a method for operating a binaural hearing system with a first hearing instrument and a second hearing instrument, wherein a first input signal is generated from ambient sound by an electro-acoustic first input transducer of the first hearing instrument, and a second input signal is generated by an electro-acoustic second input transducer of the second hearing instrument is, wherein a first instantaneous gain parameter is determined based on the first input signal, and a second instantaneous gain parameter is determined based on the second input signal.

In a hearing instrument, ambient sound is converted into an input signal by means of at least one electro-acoustic input transducer (such as a microphone), which is processed in frequency bands and thereby also amplified. This can be done, for example, in a hearing aid "in the narrower sense" in order to correct a wearer's hearing impairment by individually adapting the frequency band processing of the input signal to the wearer's audiological requirements. But other hearing instruments can also have these functions to support the wearer in everyday life.

The processed input signal is converted into an output sound signal via an output transducer of the hearing instrument, which is directed to the wearer's hearing. As part of signal processing, automatic volume control ("automatic gain control", AGC) and also dynamic compression are often applied to the input signal or to an already pre-processed intermediate signal, in which the input signal is usually only linearly amplified up to a certain limit value , and above the limit value a lower gain is applied in order to compensate for level peaks in the input signal. This is intended in particular to prevent sudden, loud sound events from leading to an output sound signal that is too loud for the wearer due to the additional amplification in the hearing aid.

However, with a binaural hearing system with two individual hearing instruments (to be worn on the left or right ear), the compression in the individual hearing instruments results in a sound signal from a slightly lateral sound source being amplified to different degrees. In particular, disturbing noises in the respective half-space (i.e. right or left) can cause the local input signal of the respective hearing instrument to be compressed more strongly by the AGC and therefore amplified less. This can lead to a loss of the natural interaural level differences required for the correct localization of sound sources. As a result, the wearer of the binaural hearing system may perceive a sound source acoustically at a location other than where the wearer sees the source.

The invention is based on the object of specifying a method for signal processing in a binaural hearing system, which enables correct acoustic localization of sound sources, particularly in conjunction with AGC and dynamic compression.

The stated object is achieved according to the invention by a method for operating a binaural hearing system with a first hearing instrument and a second hearing instrument, a first input signal being generated from ambient sound by an electro-acoustic first input transducer of the first hearing instrument, and by an electro-acoustic second input transducer of the second hearing instrument a second input signal is generated, and wherein a first instantaneous gain parameter is determined based on the first input signal, and a second instantaneous gain parameter is determined based on the second input signal.

According to the method, it is provided that a first parameter of an AGC for the first input signal and/or a second parameter of an AGC for the second input signal is adapted in such a way that as a result of said adaptation a difference between the first and the second instantaneous gain parameters is reduced, and In the first or second hearing instrument, signal processing of the first or second input signal takes place with the first or second parameter of the AGC adapted in this way. Advantageous and partly inventive embodiments are the subject of the subclaims and the following description.

The first and second hearing instruments are to be worn by the wearer on the left and right ears when the binaural hearing system is being used as intended (without this necessarily assigning the first or second hearing instrument to a specific ear).

A hearing instrument here generally includes any device that is set up to generate an electrical input signal from ambient sound and, through appropriate processing, to generate an output signal from this, which in turn is converted into an output sound signal by an output transducer. The said output sound signal is fed to the hearing of the wearer of this device. In particular, a hearing instrument is given as a headphone (e.g. as an "earplug"), a headset, data glasses with a loudspeaker, etc. A hearing instrument also includes a hearing aid in the narrower sense, i.e. a device for treating the wearer's hearing impairment in which an input signal generated from an environmental signal by means of a microphone is processed into an output signal and in particular amplified depending on the frequency band, and an output sound signal generated from the output signal by means of a loudspeaker or the like is suitable for at least partially compensating for the hearing impairment of the wearer, in particular in a user-specific manner.

An electroacoustic input transducer includes, in particular, a transducer that is designed to extract ambient sound to generate a corresponding electrical signal. In particular, when the first or second input signal is generated by the respective input converter, preprocessing can also take place, for example in the form of linear preamplification and/or A/D conversion. The correspondingly generated input signal is given in particular by an electrical signal, the current and/or voltage fluctuations of which essentially represent the sound pressure fluctuations in the air.

The first and second instantaneous amplification parameters are preferably to be determined in such a way that level peaks of the ambient sound in the respective first and second input signals are attenuated, and thereby in particular oversteering is avoided, and particularly preferably quiet sound events in the ambient sound are increased. In particular, the instantaneous gain parameters can be determined using an AGC, for example using a corresponding compression characteristic.

The first and second parameters of the AGC to be adjusted can be given directly by first and second instantaneous gain parameters; In this case, the difference can be reduced directly through the adjustment, for example by adjusting the instantaneous gain parameters to one another.

However, a compression ratio, a knee point of a compression characteristic curve, an attack time and/or a release time of a compression can also be adjusted as the first or second parameter of the AGC. In this case, the adaptation preferably has the effect that when the instantaneous gain parameters are recalculated based on the input signals processed according to the adapted AGC, there is a smaller difference in the instantaneous gain parameters. This includes in particular that natural volume differences in the ambient sound on both sides of the binaural hearing system, which are correspondingly represented in the two input signals, are preserved to a greater extent. In particular, only the first parameter of the AGC can be adjusted, or the first and second parameters of the AGC can also be adjusted, in the latter case, preferably with a “movement” of the two parameters towards each other.

The adaptation of the first or second parameter of the AGC can be carried out in particular by using the input signal of the other hearing instrument - or a transmission signal derived therefrom (which may have a lower sample rate and / or a lower dynamic range than the relevant input signal). and/or only contains some frequency bands of the input signal) - is transmitted to the "local" hearing instrument, and in a hearing instrument based on both input signals - i.e. the "local" and the "remote" (the other hearing instrument) - both instantaneous amplification parameters and accordingly the Adjustment of the relevant parameter of the AGC (or both parameters) is determined.

However, the adaptation of the first or second parameter of the AGC can also be done by determining the respective instantaneous amplification parameter locally in each hearing instrument based on the local input signal, and only transmitting this to the other hearing instrument, then using both instantaneous amplification parameters in at least the adjustment of the local parameter of the AGC is carried out in one of the hearing instruments.

The adaptation of the first or second parameter of the AGC takes place in the relevant hearing instrument, preferably on the basis of regulations that are identically defined in advance for both hearing instruments, i.e. in particular depending on the two instantaneous amplification parameters and possibly other variables, which is determined in advance by the regulation , in which way in which of the hearing instruments the respective parameter of the AGC is to be adjusted, and in what way (e.g. by lowering or by raising).

Said adjustment of either the first or the second parameter of the AGC or both parameters ensures that natural volume differences in the ambient sound are better preserved. This can be done, for example, through an adjustment of the two parameters take place together, particularly in the case of a level-related parameter of the AGC such as a compression ratio or an instantaneous gain parameter (or even a knee point of the compression characteristic curve). This fact is reflected in the fact that the difference in the instantaneous gain parameters - when these are recalculated based on the input signals processed with the adjusted parameters of the AGC - decreases. This reduced difference in volume now allows a significantly improved acoustic localization of sound sources using output signals that were generated using the adjusted parameters of the AGC from the respective input signal in the relevant hearing instrument, since the different compression, which distorts interaural level differences, can be at least partially repealed.

It proves to be advantageous if a direction of a sound source of the ambient sound is at least approximately determined based on the first input signal and the second input signal, and the adaptation of the first or second parameter of the AGC is also carried out based on the determined direction of the sound source. Here, for example, the direction of the sound source can be determined precisely to a few degrees (e.g. +/- 5° or +/- 10°), and the adjustment of the respective parameter of the AGC, which is intended or considered necessary based on the two instantaneous amplification parameters, can be made even stronger The closer the sound source is to a frontal direction of the wearer (the frontal direction lies in the plane of symmetry between the two hearing instruments when the binaural hearing system is used as intended). The further the sound source is in a lateral direction, the easier it is to localize based on the interaural level differences, so that the adjustment of the AGC on both sides can be weakened in favor of a gentler sound.

Preferably, the determination of the direction of the sound source can be combined with an analysis with regard to a useful sound signal, that is, in particular in a situation with several sound sources (some of which may be directed), an analysis can be carried out as to which sound signal from the individual sound sources is considered a useful sound signal (e.g. a speech signal) is to be considered, so that the direction of the sound source is approximately determined for this useful sound signal. The detection of the useful sound signal can be done in particular on the basis of an analysis of a modulation and/or an analysis of spectral contributions of the first and/or the second input signal.

For the approximate determination of the direction of the sound source, a focus half-space is advantageously determined, which contains the sound source, and a background half-space facing away from the focus half-space, the focus half-space and the background half-space relating to the said plane of symmetry of the binaural hearing system (in normal operation) are defined, and in particular the corresponding first or second instantaneous gain parameter of the focus half-space and / or a signal level in the focus half-space is used to adapt the first or second parameter of the AGC.

In other words, the direction of the sound source is determined only with respect to the lateral half-space in which the sound source lies. The two half-spaces are defined in relation to the plane of symmetry of the hearing instruments (as they are to be worn during intended use), with the half-space of the sound source being referred to as the focus half-space, and the remaining half-space as the background half-space. The adjustment of the first or second parameter of the AGC, and thus the coordination of the signal processing of the dynamics in both hearing instruments, then takes place depending on the signal level and/or the instantaneous amplification parameter in the half-space in which the sound source is located (i.e. in the focus half-space) . In the event that the sound level in the focus half-space is higher than the sound level in the background half-space, this can be used to determine the manner in which amplification parameters can be increased further, or in particular whether the instantaneous amplification parameter of the focus half-space (or a compression ratio there) needs to be increased for the adjustment.

• (i) $$\begin{array}{cc}\mathrm{Pout}1=\mathrm{w}1\cdot \mathrm{PC}1\mathrm{+}\left(1-\mathrm{w}1\right)\cdot \mathrm{P}1& \mathrm{mit}\mathrm{0}\le \mathrm{w}1\le 1.\end{array}$$
  
  Vergleichbares kann insbesondere auch für den zweiten Parameter erfolgen, also
• (i`) $$\begin{array}{cc}\mathrm{<figure-callout\; id="2"\; label="Pout"\; filenames="imgf0001.png"\; state="\{\{state\}\}">Pout</figure-callout>}2=\mathrm{w}2\cdot \mathrm{PC}1+\left(1-\mathrm{w}2\right)\cdot \mathrm{P}2& \mathrm{mit}0\le \mathrm{w}2\le 1,\end{array}$$
  
  dem zweiten Korrekturparameter PC2, dem ursprünglichen Wert des zweiten Parameters P2, und dem abschließend verwendeten Wert Pout2 des zweite Parameters.

In an advantageous embodiment, a first correction parameter and/or a second correction parameter is determined for the adaptation of the first or second parameter of the automatic gain control, an adapted first or second parameter being determined on the basis of a convex combination of the first parameter with the first correction parameter or the second parameter is formed with the second correction parameter. The first correction parameter or the second correction parameter preferably corresponds to a complete "one-sided" adjustment of the respective parameter, that is, for example, for the first parameter of the AGC, depending on the two instantaneous gain parameters and possibly a direction of a sound source and / or a signal level (see above), a parameter value is determined which the first parameter should preferably adopt in order to adapt the signal processing with regard to maintaining the interaural level differences only via the first parameter. This parameter value, the first correction parameter PC1, can then either be used directly for the signal processing, or can be convexly combined with the original value P1 of the first parameter (which was determined according to the first instantaneous gain parameter) to form the parameter value ultimately used in the signal processing Pout1 to get the first parameter, so

• (i) $$\begin{array}{cc}\mathrm{Pout}1=\mathrm{w}1\cdot \mathrm{PC}1\mathrm{+}\left(1-\mathrm{w}1\right)\cdot \mathrm{P}1& \mathrm{with}\mathrm{0}\le \mathrm{w}1\le 1.\end{array}$$
  
  A similar thing can also be done in particular for the second parameter, i.e
• (i`) $$\begin{array}{cc}\mathrm{<figure-callout\; id="2"\; label="Pout"\; filenames="imgf0001.png"\; state="\{\{state\}\}">Pout</figure-callout>}2=\mathrm{<figure-callout\; id="2"\; label="w"\; filenames="imgf0001.png"\; state="\{\{state\}\}">w</figure-callout>}2\cdot \mathrm{PC}1+\left(1-\mathrm{<figure-callout\; id="2"\; label="w"\; filenames="imgf0001.png"\; state="\{\{state\}\}">w</figure-callout>}2\right)\cdot \mathrm{<figure-callout\; id="2"\; label="P"\; filenames="imgf0001.png"\; state="\{\{state\}\}">P</figure-callout>}2& \mathrm{with}0\le \mathrm{<figure-callout\; id="2"\; label="w"\; filenames="imgf0001.png"\; state="\{\{state\}\}">w</figure-callout>}2\le 1,\end{array}$$
  
  the second correction parameter PC2, the original value of the second parameter P2, and the finally used value Pout2 of the second parameter.

On the one hand, the “continuous” variation of the adaptation described here can be carried out independently for each hearing instrument. This can also be done advantageously in additional dependence on a determined speech content in the input signals. If a high speech content is determined in one of the input signals, a high degree of modification of the first (and possibly the second) parameter for the adaptation can be carried out (w1 and possibly w2 is then selected close to 1), since a localization of the speech source is considered important.

On the other hand, a connection can also be established between the adjustments on both sides, in particular by functionally linking the weighting factors w1, w2, i.e. as w2 = f (w1), in particular w2 = 1 - w1. In this case, the greater the adjustment of the parameter on one side, the weaker it is on the other side.

This procedure is particularly advantageous in combination with the determination of the focus and background half-space (and the corresponding signal level of the focus half-space), since the overall extent of the adjustment can be distributed to both sides depending on the signal volume in the focus half-space.

Conveniently, the first input signal or a first transmission signal derived therefrom is transmitted from the first hearing instrument to the second hearing instrument, the first and the second instantaneous amplification parameters being determined locally in the second hearing instrument, and depending on the first and the second instantaneous amplification parameters in the second hearing instrument second parameter of the automatic gain control for signal processing of the second input signal is adjusted. This procedure is particularly advantageous if not only the two instantaneous amplification parameters but also, for example, the direction of a sound source are to be used for the adjustment.

Preferably, the second input signal or a second transmission signal derived therefrom is also transmitted from the second hearing instrument to the first hearing instrument, the two instantaneous amplification parameters being determined locally in the first hearing instrument, and the first parameter depending on the two instantaneous amplification parameters in the first hearing instrument the automatic gain control for signal processing of the first input signal is adjusted.

Conveniently, a dedicated, hard-wired circuit is used for the local determination of the first and second instantaneous amplification parameters in the second (or also in the first) hearing instrument. This means in particular that the first instantaneous amplification parameter is determined on a specially assigned hardware circuit (e.g. an ASIC) of the hearing instrument in question, and the second instantaneous amplification parameter is determined on a further hardware circuit of the same hearing instrument. Some hearing instruments already have two such dedicated circuits, one circuit being intended for an AGC of the audio signals generated in the hearing instrument, and the other circuit being intended for an AGC of a streaming signal that the hearing instrument receives, for example from a multimedia device, or from a telephone or similar. In this case, for example, an AGC ASIC originally intended for the streaming signal can be used in the second hearing instrument to determine the first instantaneous amplification parameter.

The method is preferably applied frequency band by frequency, that is, the AGC is carried out separately for each frequency band in each hearing instrument, and accordingly the relevant first or second parameter of the AGC is adapted accordingly for individual frequency bands. However, said adaptation can also be limited to individual frequency bands or in particular to a contiguous frequency range (which is particularly important for localization of the interaural level differences) from several frequency bands. This also allows the process to be implemented in an energy-efficient manner (particularly with regard to battery performance). In this case, only low frequency bands (in the said frequency range) of the first input signal are preferably transmitted to the second hearing instrument as the first transmission signal, thereby further improving the energy efficiency.

The invention further mentions a binaural hearing system with a first hearing instrument and a second hearing instrument, the binaural hearing system being set up to carry out the method described above. The binaural hearing system according to the invention shares the advantages of the method according to the invention. The advantages stated for the procedure and its further developments can be transferred analogously to the binaural hearing system.

To carry out the method, the first and second hearing instruments preferably each have a first and second input transducer for generating the first and second input signal of the method. The binaural hearing system preferably has a signal processing unit in at least one of the hearing instruments for carrying out the signal processing steps of the method, which in particular comprises at least one signal processor. Both hearing instruments particularly preferably each have such a signal processing unit.

Fig. 1a

in einer Draufsicht eine Gesprächssituation,

Fig. 1b

in einer Draufsicht die Wirkung von Dynamik-Kompression im binauralen Hörsystem auf die räumliche Hörwahrnehmung der Gesprächssituation nach Fig. 1a durch den Träger, und

Fig. 2

in einem Blockdiagramm den Ablauf eines Verfahrens für ein binaurales Hörsystem zur Verbesserung der räumlichen Hörwahrnehmung.

An exemplary embodiment of the invention is explained in more detail below with reference to drawings. Here we show schematically:

Fig. 1a

in a top view a conversation situation,

Fig. 1b

In a top view, the effect of dynamic compression in the binaural hearing system on the spatial auditory perception of the conversation situation Fig. 1a by the wearer, and

Fig. 2

in a block diagram the process for a binaural hearing system to improve spatial hearing perception.

Corresponding parts and sizes are provided with the same reference numbers in all figures.

In Fig. 1a A conversation situation is shown schematically in a top view in which a first person 1 is talking to a second person 2 who is located frontally to the first person 1 (and facing the latter). There is a third person 3 diagonally behind the first person 1, who may also say or interject things, but does not take part in the conversation between the first person 1 and the second person 2. The third person 3 could also be replaced by another directed source of noise for the following explanations.

In Fig. 1b is a schematic top view of the conversation situation Fig. 1a shown. The first person 1 is here represented by a wearer 5 of a binaural hearing system 10 according to the prior art, which includes a first hearing instrument 11 and a second hearing instrument 12. In particular, the hearing instruments 11, 12 can each be provided by a hearing aid (“in the narrower sense”). The second person 2 is according to the conversation situation Fig. 1a at the same time a conversation partner 6 of the wearer 5. The third person 3 is to be viewed as a source of interference 7 in the context of the conversation situation between the wearer 5 and his conversation partner 6.

In hearing instruments, dynamic compression is usually applied to the input signals in order to map the dynamic range, which can generally be resolved by the microphones of the hearing instruments (i.e. from the minimum recordable sound level to overdrive), into a range that is acceptable and preferably pleasant for the wearer. The lower limit for this range is then preferably given by the hearing threshold of the wearer, and the upper limit of the range is preferably given by the discomfort threshold. This is intended to ensure optimal amplification (or attenuation) in particular for all possible or realistically expected input levels at the microphones.

In the binaural hearing system 10 after Fig. 1b The dynamic compression described is applied in both hearing instruments 11, 12 independently of one another, that is, an AGC determines one for each of the two hearing instruments 11, 12 optimal amplification factor for the associated input signal. This amplification factor is usually different for both hearing instruments 11, 12, since, for example, a higher sound level is recorded on the second hearing instrument 12, which in the present case is worn by the wearer 5 on his right ear, due to the interference source 7, than on the first hearing instrument 11 (which in the present case is worn by the wearer 5 on his is worn on the left ear, and thus the source of interference 7 is shaded by the head of the wearer 5).

With dynamic compression, a higher sound level is usually assigned a lower amplification factor than a lower sound level, with the assignment being determined, for example, based on a compression characteristic curve (which describes the relationship between input level and output level). This means for the conversation situation Fig. 1a or 1b, that the amplification factor, which is determined in the first hearing instrument 11 as a first instantaneous amplification parameter G1 for application to the input signal there (or in the case of several to the input signals there), is greater than one in the second hearing instrument 12 for the purpose there Application of determined second instantaneous gain parameter G2.

As a result of the different amplification of the input signals of the two hearing instruments 11, 12 by the instantaneous amplification parameters G1, G2, the conversational contributions of the conversation partner 6 are also amplified to different degrees in the two hearing instruments 11, 12, and accordingly are also reproduced differently loudly for the wearer 5. As perceived by the wearer, this leads to the "left" contributions of the conversation partner 6 (which are recorded and processed by the first hearing instrument 11) being louder than the "right" contributions (which are recorded and processed by the second hearing instrument 12) due to G1 > G2 are processed).

However, when the input signals from the hearing instruments 11, 12 are reproduced accordingly, the interaural level differences, which are used by the ear to localize sound sources, are distorted. Through this The localization can also be distorted, ie the wearer may acoustically perceive a sound source at a different location in the room than the actual position of the sound source.

In this example according to Fig. 1b In the event that the interference source 7 emits a loud noise, the second instantaneous amplification factor G2 is reduced by the dynamic compression as a result of the increased sound level (compared to the case that the interference source 7 is silent), so the case described above occurs G1 > G2. However, since the conversation partner 6 is facing the wearer 5 in the frontal direction 14, and therefore his conversation contributions reach the wearer 5 at the same volume, these conversation contributions are reproduced louder by the first hearing instrument 11 worn on the left of the wearer 5 than by the second hearing instrument 12 worn on the right Perception of the wearer 5, this "artificial" level difference is perceived as a shadowing effect and thus as an interaural level difference due to the different amplification, so that the conversation partner 6 is no longer "heard" (i.e. perceived there) in the frontal direction 14, but in a direction 15, which is turned slightly to the left from the frontal direction 14. In Fig. 1b This is illustrated schematically using a graphical shift of the conversation partner 6 (see thick arrow) towards direction 15.

The different volumes as a result of the different instantaneous amplification parameters G1 > G2 can also have the same effect on other sound sources in the area of the wearer 5. This applies in particular to the source of interference 7. In realistic situations, such a source of interference may also be a danger to the wearer 5 (e.g. from an approaching vehicle in traffic), which is why a spatially distorted perception is also a concern for safety reasons.

To do this based on Fig. 1b To solve the problem described, a method is provided for the binaural hearing system 10, the process of which is based on Fig. 2 shown in a block diagram. The first hearing instrument 11 has an electroacoustic first input transducer M1, which is set up to to generate a first input signal E1 from ambient sound 18, and which in the present case is provided by a microphone. The first hearing instrument 11 can have a further input transducer (not shown), by means of which a further input signal is generated from ambient sound 18, so that directional processing of the local input signals can take place in the hearing instrument 11.

The second hearing instrument 12 has an electroacoustic second input transducer M2, which is set up to generate a second input signal E2 from the ambient sound 18, and which in the present case is also provided by a microphone. The second hearing instrument 12 can also have a further input transducer (not shown) for local directional processing.

Based on the first input signal E1, a first transmission signal T1 is generated, which is transmitted from the first hearing instrument 11 to the second hearing instrument 12. The first transmission signal can be generated, for example, from a frequency range of contiguous frequency bands of the first input signal E1. In the above-mentioned case of directional processing of two input signals in the first hearing instrument 11, the first transmission signal can also be given by the resulting directional signal (or frequency bands thereof). However, the first transmission signal T1 can also be given directly by the complete first input signal E1. In an analogous manner, a second transmission signal T2 is generated based on the second input signal E2, which is transmitted from the second hearing instrument 12 to the first hearing instrument 11.

In the first hearing instrument, a first instantaneous gain parameter G1 for the first input signal E1 is determined frequency band by frequency band using the first input signal E1 by a first local AGC 21-L. This is preferably to be determined in such a way that for the ambient sound 18 represented in the first input signal E1 by the first instantaneous amplification parameter G1 with regard to the dynamic range of the hearing instrument 11 and the hearing of the wearer 5 optimal amplification is achieved. Likewise, in the first hearing instrument 11, a second instantaneous gain parameter G2 for the second input signal E2 is determined locally by a first remote AGC 21 -R based on the second transmission signal T2 in frequency bands according to the same rules as the first instantaneous gain parameter G1 based on the first input signal E1. The second instantaneous amplification parameter G2 thus forms the optimal amplification for the second input signal E2 in the respective frequency band with regard to the dynamics and hearing ability of the wearer 5.

It should be taken into account here that in the present example the second input signal E2 is identical to the second transmission signal T2 in the relevant frequency bands (i.e. in those in which T2 is different from zero). In the case (not shown) of two input signals per hearing instrument, which are preprocessed locally to form corresponding directional signals, the two input signals E1, E2 are preferably replaced by the said locally generated directional signals. This means in particular that the instantaneous amplification parameters G1, G2 are preferably generated frequency band by frequency from the corresponding directional signals (in particular the respective directional signal, possibly limited to some frequency bands of the same, also serves as a transmission signal).

In an analogous manner, the second instantaneous amplification parameter G2 is determined in the second hearing instrument 12 in frequency bands by a second local AGC 22-L from the second input signal E2 and the first instantaneous amplification parameter G1 is determined by a second remote AGC 22-R from the first transmission signal T1. As a result of the frequency band-wise identical signal components, which are used in both hearing instruments 11, 12 to determine the first and second instantaneous amplification parameters G1, G2, as well as as a result of the identical algorithms in the local first AGC 21-L and the remote second AGC 22-R (as well as in the remote first AGC 21-R and the local second AGC 22-L) in both hearing instruments 11, 12 the respectively determined first instantaneous amplification parameters G1 are identical to one another (and the respectively determined second instantaneous amplification parameters G2 are identical to one another).

Furthermore, in a first source determination Q1 of the first hearing instrument 11, at least approximately a direction 25 of a sound source 30 in the ambient sound 18 is determined frequency band by frequency based on the first input signal E1 and the second transmission signal T2. This approximate determination can, for example, determine a polar angle (possibly accurate to 5°, 10° or similar) of the sound source with respect to the frontal direction 14, or even just a half-space with respect to a plane of symmetry 28 of the binaural hearing system 10 containing the frontal direction 14 , in which the sound source 30 lies. This half-space in question is referred to here as the focus half-space 31. Analogously, the at least approximate direction 25 is also determined frequency band by frequency in a second source determination Q2 of the second hearing instrument 12 based on the second input signal E2 and the first transmission signal T1. Since the same signal components in the frequency bands are used in the first and second hearing instruments 11, 12 (i.e. that the respective input signal E1 or E2 is identical to its transmission signal T1 or T2 in the frequency bands used), Q1, Q2 determines the identical direction 25. In the case (not shown) that there are two input signals in each hearing instrument 11, 12, each of which is preprocessed locally to form corresponding directional signals, the frequency band-wise directional signals are preferably supplied to the first and second source determinations Q1, Q2.

Based on the direction 25, the focus half-space 31 in which the sound source 30 is located is determined in each of the two hearing instruments 11, 12 (if this has not already been done by the approximate determination of the direction 25), and, as a result, the the half-space opposite the focus half-space 31, which should be referred to here as the background half-space 32.

Based on the first and second instantaneous amplification parameters G1, G2 and based on the knowledge of the focus half-space 31, a first adjustment 41 of a first parameter P1 of the AGC is now carried out in the first hearing instrument 11, which is used locally in the first hearing instrument 11 for signal processing (ie, in the present exemplary embodiment, no "off-site" parameter of the second hearing instrument 12 is adapted in the first hearing instrument 11). Additionally or alternatively, a second adjustment 42 of a second parameter P2 of the AGC is carried out in the second hearing instrument 12, which is used locally in the first hearing instrument 12 for signal processing.

In the present case, the first parameter P1 is given by the first instantaneous gain parameter G1, and the second parameter P2 by the second instantaneous gain parameter G2. In the event that G1 < G2 (e.g. if the sound source 30 in the focus half-space 31 leads to a higher sound level due to shading effects than in the background half-space 32, and there is no excessively loud source of interference there), for example . The adaptation consists of simply using the value of the first instantaneous gain parameter G1 for the second parameter P2 through the second adaptation 42, so that the amplification of the input signals E1, E2 in both hearing instruments 11, 12 is the same. Such a lowering merely dampens additional background noise in the background half-space 32. Conversely, if G1 > G2 applies (e.g. as a result of a loud source of interference in the background half-space 32 with a simultaneous useful signal from the sound source 30), then an adaptive adjustment of the parameters P1, P2 (i.e. the two instantaneous amplification parameters G1, G2) can be carried out depending on additional speech recognition ( not shown) of the input signals E1, E2 (or the transmission signals T1, T2). In short signal sections (e.g. frames or other time bins of a suitable length) with speech components, the adaptation can be suspended if necessary in order to prevent G2 from being increased (increasing the noise background) or G1 (and thus the speech signal from being lowered). ) the speech signal would become incomprehensible. The adjustment (e.g. by adjusting the instantaneous gain parameters G1, G2 for the values of the parameters P1, P2) is then limited to the cases in which there is no speech signal.

Other previously described types of adjustments - in particular both parameters P1 and P2 simultaneously - can also be carried out.

The first input signal E1 is then further processed in the first hearing instrument 11 with the correspondingly adapted first parameter P1 to a second output signal Ou1, the second input signal E2 in the second hearing instrument 12 with the correspondingly adapted second parameter P2 to a second output signal Ou2 (where, as mentioned, if necessary, the adjustment only has a non-trivial effect on one of the two parameters). The two output signals Ou1, Ou2 can then be subjected to further signal processing steps, not shown in more detail (e.g. additional suppression of noise and/or acoustic feedback, etc.), and are then transmitted by an electroacoustic first and second output transducer L1 and L2, respectively converted into a first and second output sound signal 51 and 52, respectively.

Although the invention has been illustrated and described in detail by the preferred embodiment, the invention is not limited by the examples disclosed and other variations may be derived therefrom by those skilled in the art without departing from the scope of the invention.

Reference symbol list

1

first person

2

second person

3

third person

5

Wearer (of the binaural hearing system)

6

interlocutor

7

Source of interference

10

binaural hearing system

11

first hearing instrument

12

second hearing instrument

14

Frontal direction

15

Direction

18

Ambient noise

21-L/-R

first local or remote AGC

22-L/-R

second local or remote AGC

25

Direction (of the sound source)

28

plane of symmetry

30

Sound source

31

Focus half space

32

Background half space

41

first adjustment

42

second adjustment

51

first output sound signal

52

second output sound signal

E1, E2

first or second input signal

G1, G2

first and second instantaneous gain parameters

L1, L2

(electroacoustic) first or second output transducer

M1, M2

(electroacoustic) first or second input transducer

Ou1, Ou2

first or second output signal

P1, P2

first or second parameter (of an AGC)

Q1, Q2

first or second source identification

T1, T2

first or second transmission signal

Claims (12)

Method for operating a binaural hearing system (10) with a first hearing instrument (11) and a second hearing instrument (12), wherein a first input signal (E1) is generated from ambient sound (18) by an electro-acoustic first input transducer (M1) of the first hearing instrument (11), and a second input signal (E2) by an electro-acoustic second input transducer (M2) of the second hearing instrument (12). ) is produced, wherein a first instantaneous gain parameter (G1) is determined based on the first input signal (E1), and a second instantaneous gain parameter (G2) is determined based on the second input signal (E2), wherein a first parameter (P1) of an automatic gain control for the first input signal (E1) and/or a second parameter (P2) of an automatic gain control for the second input signal (E2) is adjusted in such a way that as a result of said adjustment (41, 42) a difference between the first and second instantaneous gain parameters (G1, G2) is reduced, and wherein in the first or second hearing instrument (11, 12), signal processing of the first or second input signal (E1, E2) takes place with the first or second parameter (P1, P2) of the automatic gain control adapted in this way. Method according to claim 1,
wherein the first or second instantaneous gain parameter (G1, G2) is adjusted as the first or second parameter (P1, P2) of the automatic gain control. Method according to claim 1 or claim 2,
wherein a compression ratio and/or a knee point of a compression characteristic curve and/or an attack time and/or a release time of a compression are adjusted as the first or second parameter (P1, P2) of the automatic gain control. Method according to one of the preceding claims,
wherein a direction (25) of a sound source (30) of the ambient sound (18) is at least approximately determined based on the first input signal (E1) and based on the second input signal (E2), and wherein the adaptation (41, 42) of the first or second Parameters (P1, P2) of the automatic gain control also take place based on the determined direction (25) of the sound source (30). Method according to claim 4,
wherein the adjustment (41, 42) of the first or second parameter (P1, P2) of the automatic gain control occurs more strongly the closer the determined direction (25) of the sound source (30) is to a frontal direction of the binaural hearing system (10). Method according to claim 4, for the approximate determination of the direction (25) of the sound source (30), a focus half-space (31) is determined, which contains the sound source (30), and a background half-space (32) facing away from the focus half-space (31), wherein the focus half-space (31) and the background half-space (32) are defined with respect to a plane of symmetry (28) of the binaural hearing system (10), and wherein for the adaptation (41, 42) of the first or second parameter (P1, P2) of the automatic gain control, in particular the corresponding first or second instantaneous gain parameter (G1, G2) of the focus half-space (31) or a signal level in Focus half space (31) is used. Method according to one of the preceding claims, wherein a first correction parameter and/or a second correction parameter is determined for the adaptation (41, 42) of the first or second parameter (P1, P2) of the automatic gain control, wherein an adapted first or second parameter (P1, P2) is formed based on a convex combination of the first parameter (P1) with the first correction parameter or of the second parameter (P2) with the second correction parameter. Method according to one of the preceding claims, wherein the first input signal (E1) or a first transmission signal (T1) derived therefrom is transmitted from the first hearing instrument (11) to the second hearing instrument (12), wherein the first and second instantaneous amplification parameters (G1, G2) are determined locally in the second hearing instrument (12), and wherein depending on the first and second instantaneous gain parameters (G1, G2) in the second hearing instrument (12), the second parameter (P2) of the automatic gain control for signal processing of the second input signal (E2) is adapted. Method according to claim 8,
wherein a dedicated, hard-wired circuit is used for the local determination of the first and second instantaneous amplification parameters (G1, G2) in the second hearing instrument (12). Method according to one of the preceding claims,
whereby the method is applied frequency band by frequency. Method according to claim 10 in conjunction with one of claims 8 or 9,
wherein only low frequency bands of the first input signal (E1) are transmitted to the second hearing instrument (12) as the first transmission signal (T1). Binaural hearing system (10) with a first hearing instrument (11) and a second hearing instrument (12),
wherein the binaural hearing system (10) is set up to carry out the method according to one of the preceding claims.

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