Method for directional signal processing for a hearing system

Abstract

The invention describes a method for directional signal processing for a hearing system (HS) with at least one first hearing instrument (H1), wherein a first input signal (E1) is generated from ambient sound (Us) by an electroacoustic first input transducer (M1) of the first hearing instrument (H1), and a second input signal (E2) is generated by an electroacoustic second input transducer (M2) of the hearing system (HS), wherein a number (#N) of conversation partners (G1-3, Gj) of a wearer (T1) of the hearing system (HS) is determined based on the first input signal (E1) and based on the second input signal (E2), wherein, depending on the determined number (#N) of conversation partners (G1-3, Gj), at least one compression and / or directional microphone technique and / or noise reduction is modified in the processing of the first input signal (E1) and / or the second input signal (E2), thereby generating a first output signal (Out1). becomes.

Description

[0001] The invention relates to a method for directional signal processing for a hearing system having at least one first hearing instrument, wherein a first input signal is generated from an ambient sound by an electroacoustic first input transducer of the first hearing instrument, and a second input signal is generated by an electroacoustic second input transducer of the hearing system, wherein a first output signal is generated depending on a processing of the first input signal and / or the second input signal.

[0002] One of the most challenging situations in a hearing aid for treating hearing loss is supporting a hearing-impaired person in a conversation with multiple potential conversation partners (e.g., a so-called "cocktail party" listening situation). In this case, several useful signals, which are present in the conversation partners' contributions, must usually be separated from a background noise, which in turn may contain additional, possibly undefined or fragmented speech signals, and which, moreover, usually lacks clearly defined sources.

[0003] In a one-on-one conversation, a hearing-impaired person can usually concentrate fully on the conversation with the sole interlocutor, allowing them to pay particular attention to non-verbal communication elements such as the interlocutor's gestures and / or facial expressions, as well as to direct interaction with the interlocutor (e.g., short, mutual interjections, etc.). A hearing aid can effectively support this conversation by emphasizing the speech contributions of the sole interlocutor.

[0004] However, when there are multiple conversation partners, the support provided by a hearing aid is often difficult simply because the individual contributions cannot be adequately highlighted against background noise, or because such highlighting can result in an unnatural-sounding and thus unpleasant background. Furthermore, one of the conversation partners is often not positioned directly in front of the hearing-impaired person, meaning that non-verbal communication elements are not perceived to the same extent as they would be in a direct conversation. This, as well as a potential lack of synchronization of so-called "turntaking," i.e., the order of speaking, can lead to considerable strain for a hearing-impaired person, causing them to increasingly withdraw from communication or even leave the conversation altogether, as they are increasingly unable to follow it.

[0005] The invention is therefore based on the object of specifying a method for a hearing instrument by means of which support for a hearing-impaired person can be achieved as precisely as possible in complex conversation situations, while the background should remain as natural as possible.

[0006] The stated object is achieved according to the invention by a method for directional signal processing for a hearing system with a first hearing instrument, wherein a first input signal is generated from ambient sound by an electroacoustic first input transducer of the first hearing instrument, and a second input signal is generated by an electroacoustic second input transducer of the hearing system, wherein a number of conversation partners of a wearer of the hearing system is determined on the basis of the first input signal and the second input signal, and wherein, depending on the determined number of conversation partners, at least one compression and / or a directional microphone and / or a noise suppression in the processing of the first input signal and / or the second input signal is modified, and a first output signal is thereby generated.Advantageous and partly inventive embodiments are the subject of the dependent claims and the following description.

[0007] A hearing instrument generally includes any device that is configured to generate an electrical signal from an ambient signal using an input transducer, process this signal into an output signal, and generate an output sound signal from the output signal and feed it to the ear of a wearer of this device. A hearing instrument particularly includes headphones (e.g., earbuds), 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 a wearer with a hearing impairment, in which one or more input signals are processed into the said output signal depending on the wearer's audiological requirements and are amplified, in particular, depending on the frequency band, so that the sound produced from the output signal is transmitted via a loudspeaker or similar.The output sound signal generated is suitable for at least partially compensating for the wearer's hearing impairment, particularly in a user-specific manner.

[0008] An electroacoustic input transducer includes, in particular, any device configured to generate a corresponding electrical signal from a sound signal. In particular, preprocessing can also be performed during the generation of the first or second input signal by the respective input transducer, e.g., in the form of linear preamplification and / or A / D conversion. The correspondingly generated input signal is provided, in particular, by an electrical signal whose current and / or voltage fluctuations essentially represent the sound pressure fluctuations in the air.

[0009] The hearing system can, on the one hand, comprise only the first hearing instrument. In this case, the second input transducer is also located in the first hearing instrument. The number of conversation partners can then be determined, in particular, using (monaural) directional microphones and / or other directional signal processing, with additional spectral analyses of the input signals being used.

[0010] Preferably, however, the hearing system is designed as a binaural hearing system comprising the first hearing instrument and a second hearing instrument, which are to be worn on one ear and the other, respectively, with the second input signal being generated by the second input transducer of the second hearing instrument. In this case, the number of conversation partners can be achieved in particular by directional processing of the two input signals, for example, by generating intermediate signals with different directional characteristics based on the two input signals using directional microphones, and the intermediate signals are used to detect the conversation partners.

[0011] Typically, signal processing in a hearing instrument, especially in a hearing aid for hearing loss, occurs depending on classified "listening situations." These listening situations are standardized types of possible acoustic environments, which are determined and classified based on certain measurable characteristics in the hearing instrument, such as signal-to-noise ratio (SNR), speech content, speech intelligibility index (SII), and similar.

[0012] However, this classification can lead to complex listening situations with multiple conversation partners in the presence of background noise (which may also contain speech signals that are irrelevant to the wearer), on the one hand, either too aggressive noise suppression and / or directional microphones are used, resulting in an unnatural sound impression, and in addition, a conversation partner standing to the side may not be picked up by the directional microphones at all, or on the other hand, individual conversation contributions are not sufficiently highlighted.

[0013] The invention solves this problem by examining the acoustic conditions of a situation to determine how many conversation partners are engaged in a conversation with the wearer of the hearing instrument or hearing system. In particular, different signal processing is explicitly applied for the case of one conversation partner on the one hand and for the case of two or more conversation partners on the other. Preferably, even with two or more conversation partners, the signal processing is applied in a case-by-case manner, thus, in particular, distinguishing between the case of two conversation partners and the case of three or more conversation partners.

[0014] Modifying compression and / or directional microphones and / or noise suppression in the processing of the first input signal and / or the second input signal particularly includes deriving an intermediate signal from the first input signal and / or the second input signal, in particular also using the signal components of further input signals, and adjusting the compression or noise suppression for this intermediate signal depending on the number of conversation partners. In particular, the intermediate signal can also be provided by a directional signal.

[0015] This specifically involves applying a specific value to a compression or noise suppression or directional microphone parameter when only one conversation partner is detected and an otherwise given acoustic situation. A different, corresponding value is assigned to the parameter when two or more conversation partners are detected. The first output signal is then preferably generated from the intermediate signal or formed directly by it.

[0016] When determining the number of conversation partners, the speech activity of a determined conversation partner is preferably dynamically taken into account such that after a predetermined period without speech activity, e.g., after at least 100 ms, preferably 1 s, and particularly preferably 3 s, as well as a maximum of three minutes, preferably a maximum of one minute, and particularly a maximum of 10 s, a person is no longer counted as a conversation partner. Thus, preferably after such a period without a corresponding speech contribution from a person who was previously considered a conversation partner, the number of conversation partners is reduced by 1 (unless a new conversation partner joins).

[0017] Preferably, a first half-space signal and a second half-space signal are generated using directional microphones based on the first input signal and the second input signal, wherein the compression and / or directional microphones and / or noise suppression are applied to the first or second half-space signal depending on the determined number of conversation partners, and / or are applied separately for each of the two half-space signals. This particularly comprises the respective half-space signal largely, preferably maximally, attenuating sound from one half-space and accordingly only making significant contributions to sound from the other half-space (which is therefore complementary to the aforementioned half-space). In particular, the first half-space, for which the first half-space signal makes significant contributions, can be complementary to the second half-space, which makes significant contributions to the second half-space signal (but not to the first half-space signal).In particular, the first and / or second half-space signal can also be generated using one or more intermediate signals, wherein the or each intermediate signal is generated using the first and second input signals, in particular by means of directional microphones. The half-space signals represent an advantageous type of preprocessing, to which further signal processing can then be applied, depending on the number of conversation partners, to improve the intelligibility of the conversation partners.

[0018] Preferably, the first half-space signal has a directional characteristic that maximally attenuates a rear half-space, and / or the second half-space signal has a directional characteristic that maximally attenuates a front half-space, wherein the front and rear half-spaces are defined with respect to a frontal direction of the wearer. This means, in particular, that the aforementioned first half-space (for which only the first half-space signal makes significant contributions) is defined by the front half-space, defined with respect to the frontal direction of the wearer (when the hearing system is worn as intended), and the aforementioned second half-space (for which only the second half-space signal makes significant contributions) is defined by the rear half-space.

[0019] Advantageously, a first auxiliary input signal is generated from the ambient sound by a first auxiliary input transducer of the first hearing instrument, and preferably a second auxiliary input signal is generated by a second auxiliary input transducer of the second hearing instrument. The number of conversation partners of the wearer of the binaural hearing system is additionally determined based on the first auxiliary input signal and / or the second auxiliary input signal. This means, in particular, that the binaural hearing system generates two input signals in each of the two hearing instruments, to which local preprocessing (through directional microphones) can be applied if necessary.The number of conversation partners is determined at least in one (in the first) hearing instrument based on at least the two local input signals (the first input signal and the first auxiliary input signal) and at least one input signal (the second input signal), and preferably based on both input signals (i.e. also the second auxiliary input signal) of the other hearing instrument (the second hearing instrument). For this purpose, in particular the respective input signals (or an intermediate signal derived from them, which may optionally be generated by means of local directional microphones) can be transmitted to the respective other hearing instrument. The procedure described here allows a particularly precise determination, in particular of the position of a conversation partner, to be carried out using the auxiliary input signals.

[0020] Preferably, the first output signal is also generated based on the first input signal and the first auxiliary input signal, and in particular based on the second input signal and / or the second auxiliary input signal. This includes, in particular, the first output signal incorporating the signal components of the same input signals that are also used to determine the number of conversation partners.

[0021] Advantageously, a speech component in the ambient sound is monitored based on the first and second input signals, and preferably also based on the first and / or second auxiliary input signals. A spatial analysis is carried out based on the first and second input signals to determine the number and position of the conversation partners. For this purpose, in particular, an angular direction of a sound source of the ambient sound is determined. The presence and position of an individual speaker are determined based on the speech component and the angular direction. This is particularly advantageous as a preliminary stage for determining the number of conversation partners. In particular, a spectral analysis is applied to individual speech components in order to achieve a better differentiation between different speakers.

[0022] Advantageously, in a further step, based on the first and second input signals, and preferably also based on the first and / or second auxiliary input signals, the length of a conversation contribution and / or an overlap of a conversation contribution with a conversation contribution of the wearer is determined for said individual speaker, and from this, the individual speaker is identified as a conversation partner of the wearer. This makes it possible, in particular, to filter out speakers who are involved in other conversations in a confusing acoustic environment and are therefore not conversation partners of the wearer.

[0023] For practical purposes, the presence and position of each speaker are determined only in one half-space, which corresponds to one of the two half-space signals. This allows for pre-filtering of the acoustic environment, particularly for detection in the front half-space.

[0024] It also proves advantageous to track a change in the position of an individual speaker. This can be achieved, for example, by updating the position sufficiently frequently. By tracking their position, a conversation partner who has been recognized once can remain identified as such without the need for a continuous, comprehensive analysis of their spectral frequency components in the speech portion or the overlap of their conversational contributions.

[0025] In an advantageous embodiment, if only one call partner is detected, a predetermined first value of an amplification parameter is applied to an intermediate signal generated from the first and / or second input signal. If exactly two call partners are detected, a second value of the amplification parameter is applied to the intermediate signal, which is 2 dB to 5 dB lower than the first value. If exactly three call partners are detected, a third value of the amplification parameter is applied to the intermediate signal, which is 4 dB to 10 dB lower than the first value. The intermediate signal can in particular be provided by a directional signal and in particular be included in the first output signal, wherein additional frequency-band-dependent amplification and / or compression is optionally applied to the intermediate signal.

[0026] Preferably, quantitatively different signal processing is used for one, two, or three call partners, in particular by applying greater noise suppression the more call partners are present. For three or more call partners, the signal processing can remain the same.

[0027] Preferably, the gain parameter used is a gain and / or a noise gain and / or a background gain offset for the second half-space signal, which has a directional characteristic that maximally attenuates a front half-space. This means that sound from the rear half-space is increasingly attenuated as the number of conversation partners increases.

[0028] The invention further relates to a hearing system with at least one first hearing instrument, wherein the hearing system is configured to perform the method described above. Preferably, the hearing system is configured as a binaural hearing system that further comprises a second hearing instrument.

[0029] The hearing system according to the invention shares the advantages of the method according to the invention. The advantages stated for the method and its further developments can be applied analogously to the hearing system.

[0030] An embodiment of the invention is explained in more detail below with reference to the accompanying drawings, each of which shows schematically: Fig. 1 shows a block diagram of a binaural hearing system with two hearing aids, Fig. 2 shows a wearer of the binaural hearing system according to Fig. 1and its surroundings, and Fig. 3 in a block diagram the sequence of a method for signal processing for the binaural hearing system according to Fig. 1 depending on the conversation partners in the area Fig. 2 .

[0031] Corresponding parts and sizes are provided with the same reference numerals in all figures.

[0032] In Figure 1A block diagram of a binaural hearing system HS with a first hearing instrument H1 and a second hearing instrument H2 is shown schematically. The two hearing instruments H1, H2 are configured as a first and second hearing aid HG1, HG2, respectively, which are designed and provided to treat a hearing impairment of the wearer of the binaural hearing system HS. The first hearing aid HG1 has a first input transducer M1 and a first auxiliary input transducer Mh1, while the second hearing aid HG2 has a second input transducer M2 and a second auxiliary input transducer Mh2. The said input transducers are each provided by corresponding microphones.

[0033] The first input transducer M1 and the first auxiliary input transducer Mh1 are each configured to generate a first input signal E1 and a first auxiliary input signal Eh1, respectively, from an ambient sound Us. Accordingly, the second input transducer M2 and the second auxiliary input transducer Mh2 are each configured to generate a second input signal E2 and a second auxiliary input signal Eh2, respectively, from the ambient sound Us.

[0034] The first hearing aid HG1 has a first control unit St1, which processes the first input signal E1 and the first auxiliary input signal Eh1 in a manner described below to form a first output signal Out1, which is converted into a first output sound signal Sout1 by a first output transducer L1 of the first hearing aid HG1. The first output transducer L1 is provided by a loudspeaker. However, the first output transducer L1 could just as easily be provided by a bone conduction earpiece. Correspondingly, the second hearing aid comprises a second control unit St2, which processes the second input signal E2 and the second auxiliary input signal Eh2 to form a second output signal Out2. The second output signal Out2 is converted into a second output sound signal by a second output transducer (not shown) of the second hearing aid HG2.

[0035] In Figure 2is a top view of a carrier T1 of the binaural hearing system HS according to Figure 1shown. Several people P1-P5 are located in the wearer's surroundings, whereby other, not shown people may also be present, particularly outside the section shown. Some of the people P1-P5 are in conversation with each other, or with the wearer T1, whereby additional background noises not yet shown in detail (e.g. from conversations that are more distant or can no longer be localized, or from traffic or other ambient noise) may also be present. The people P1, P2, P3, who are located in a front half-space R1 with respect to a frontal direction F of the wearer T1, are engaged in a conversation with the wearer T1 and are therefore to be regarded as his conversation partners G1, G2, G3. The person P4, who is also standing in the front half-space R1 of the wearer T1, is taking part in another conversation.The person P5 is standing in the rear half-space R2 of the carrier T1, which is complementary to the front half-space R1, and consequently is not taking part in the conversation with the carrier, but is involved in another conversation (the corresponding conversation partners of the persons P4 and P5 are in . Figure 2 not shown).

[0036] The multitude of conversations surrounding him (here, for example, the conversations of people P4 and P5) as well as additional background noise make it difficult for wearer T1 to follow his own conversation with his conversation partners G1-G3. He may have to concentrate more intensely, is less able to pay attention to the non-verbal communication elements (e.g., gestures, facial expressions) of the conversation partners G1-G3, and is therefore potentially limited in his full participation in the conversation.

[0037] To solve this problem, a method is proposed which uses a Figure 3 shown block diagram.

[0038] As in Figure 1 As described above, the first input signal E1 and the first auxiliary input signal Eh1 are generated from the ambient sound Us in the first hearing aid HG1, and the second input signal E2 and the second auxiliary input signal Eh2 are generated in the second hearing aid HG2. The ambient sound Us includes the conversational contributions of persons P1-P5, as well as other localized and / or diffuse background noises.

[0039] Based on the input signals E1, E2, Eh1, Eh2, the number of call partners of the carrier T1 for the Figure 2The situation described is determined. For this purpose, the input signals E2, Eh2 generated in the second hearing aid and / or a second intermediate signal Z2 generated by these signals through local preprocessing are transmitted from the second hearing aid HG2 to the first hearing aid HG1 using suitable communication devices (not shown) of the two hearing aids HG1, HG2. In the present exemplary embodiment, only the second intermediate signal Z2 is transmitted; the input signals E2, Eh2 of the second hearing aid HG2 are not transmitted to the first hearing aid.

[0040] Based on the first input signal E1 and the first auxiliary input signal Eh1, as well as on the second input signal E2 and the second auxiliary input signal Eh2 (particularly in the form of the second intermediate signal Z2), the angular directions αj of individual sound sources SQj and the speech components SpA are determined. For this purpose, a directional signal with a variable maximum direction (lobe-shaped directional signal) or with a variable minimum direction (notch-shaped directional signal) can be generated from the aforementioned input signals E1, Eh1, E2, Eh2 or Z2, for example. In particular, the angular direction αj of a sound source SQj can first be determined, and then the speech component SpA for this sound source SQj can be determined in order to detect whether the sound source SQj is a speaker. This determines the presence and position of individual speakers in the environment of the wearer T1.In particular, individual speakers can be distinguished from one another by a spectral analysis of the speech components SpA.

[0041] In order to determine whether speech contributions Bj of a speaker identified and localized as described belong to a conversation with carrier T1, and whether the speaker is therefore a conversation partner Gj of carrier T1, the lengths Dj of the speech contributions Bj are determined, as well as a respective overlap ULj of the speech contributions Bj with speech contributions BT of carrier T1. Up to a suitably selected limit, an overlap ULj can exist for a conversation partner Gj, since individual interjections by the participants are to be expected again and again in a conversation, which also continually redefines the order of speaking. However, if the overlap ULj exceeds this limit, which can in particular also be selected depending on a length Dj of the respective speech contribution Bj, it can be assumed that this speech contribution Bj is intended for a conversation other than the one with carrier T1. Accordingly, the speaker is not counted as a conversation partner.In particular, once a conversation partner gj has been recognized and located, his position can be tracked (not shown) in order to avoid having to carry out the resource-intensive verification of his speech contributions Bj again.

[0042] In the manner described, the individual conversation partners Gj of wearer T1 can be identified, and thus also their number #N. In particular, the determination of the number of conversation partners can be restricted to the front half-space R1 (for example, by identifying and localizing sound sources SQj only in the front half-space R1).

[0043] Depending on the said number #N of conversation partners Gj, the first output signal Out1 is now generated on the basis of the above-mentioned input signals E1, Eh1, E2, Eh2. For this purpose, a first half-space signal SR1 is generated in the first hearing aid HG1 (i.e. in particular in the first control unit St1) on the basis of the said input signals E1, Eh1, E2, Eh2 (the input signals E2, Eh2 of the second hearing aid HG2 can in particular be pre-processed to form the second intermediate signal Z2) by means of a directional microphone BF. This first half-space signal SR1 only makes a significant contribution to sound sources SQj from the front half-space R1, which in this case is defined as a first half-space, and largely completely attenuates sound sources SQj from the rear half-space R2, which in this case is defined as a second half-space (whereby a narrow transition area, e.g.approximately 10°-20° angular expansion exists between the first and second half-spaces, whose sound sources are incorporated to a lesser extent into the first half-space signal SR1). The first half-space signal SR1 is thus a directional signal into which the signal components of the input signals E1, Eh1, E2, Eh2 are incorporated (if necessary after local preprocessing to form the second intermediate signal Z2, and if necessary also to form a first intermediate signal from the input signals E1, Eh1).

[0044] In a similar way, a second half-space signal SR2 is generated from the said input signals E1, Eh1, E2, Eh2 by means of the directional microphone BF, which second half-space signal SR2 only provides significant contributions for sound sources SQj from the front half-space R2, and largely completely attenuates sound sources SQj from the front half-space R1 (whereby a narrow transition area may also exist between the first and the second half-space, the sound sources of which are incorporated to a lesser extent into the second half-space signal SR2).

[0045] Depending on the number #N of conversation partners Gj, a gain value V1, V2 or V3 for #N=1, #N=2 or #N ≥ 3 is now applied to the second half-space signal SR2 (i.e., in the described dependency, a specific value for a gain applied to the second half-space signal SR2). This can be done in a frequency band-dependent manner. The thus amplified second half-space signal SR2 is combined with the first half-space signal SR1, and from this the first output signal Out1 is generated (any frequency-dependent amplification or compression of the combined signal from SR1 and SR2 is still possible, but is Figure 3 not shown separately).

[0046] Analogous to the described procedure, compression can also be applied depending on the number #N of conversation partners Gj, for example, by applying successively shorter attack constants and / or a larger compression ratio to a first intermediate signal Z1 derived from the first input signal E1 and the first auxiliary input signal Eh1, or to the first half-space signal SR1, for one, two, and three (or even more) conversation partners. Further signal processing options depending on the number #N of conversation partners Gj when generating the first output signal Out1 are conceivable.

[0047] Although the invention has been illustrated and described in detail by the preferred embodiment, the invention is not limited to the disclosed examples and other variations may be derived therefrom by those skilled in the art without departing from the scope of the invention. List of reference symbols

[0048] BFRiflective microphone BjSpeech contribution BTSpeech contribution (of the wearer) DjLength (of the speech contribution Bj) E 1 / 2first / second input signal Eh1 / 2first / second auxiliary input signal FFrontal direction G1-3, GjConversation partner H1 / 2first / second hearing instrument HG1 / 2first / second hearing aid HSbinaural hearing system L1first output transducer M1 / 2first / second input transducer Mh1 / 2first / second auxiliary input transducer Out1 / 2first / second output signal P1-5Person R1 / 2anterior / posterior half-space Sout1first output sound signal SpASpeech component SQjSound source SR1 / 2first / second half-space signal St1 / 2first / second control unit T1Carrier UsAmbient sound ULjOverlap V1-3Gain value Z2second intermediate signal #NNumber of conversation partners αJAngle direction

Claims

1. Method for directional signal processing for a hearing system (HS) with at least one first hearing instrument (H1), wherein a first input signal (E1) is generated from an ambient sound (Us) by an electroacoustic first input transducer (M1) of the first hearing instrument (H1), and a second input signal (E2) is generated by an electroacoustic second input transducer (M2) of the hearing system (HS), wherein a number (#N) of conversation partners (G1-3, Gj) of a wearer (T1) of the hearing system (HS) is determined based on the first input signal (E1) and the second input signal (E2), wherein, depending on the determined number (#N) of conversation partners (G1-3, Gj), at least one compression and / or directional microphone and / or noise suppression in the processing of the first input signal (E1) and / or the second input signal (E2) is modified, and thereby a first output signal (Out1) is generated.

2. Method according to claim 1, wherein the hearing system (HS) is a binaural hearing system with the first hearing instrument (H1) and a second hearing instrument (H2), and wherein the second input signal (E2) is generated by a second input transducer (M2) of the second hearing instrument (H2).

3. Method according to claim 1 or claim 2, wherein a first half-space signal (SR1) and / or a second half-space signal (SR2) are generated by means of directional microphones on the basis of the first input signal (E1) and the second input signal (E2), and wherein the compression and / or directional microphones and / or noise suppression are applied to the first or second half-space signal (SR1, SR2) as a function of the determined number (#N) of conversation partners (G1-3, Gj), and / or are applied separately for each of the two half-space signals (SR1, SR2).

4. The method according to claim 3, wherein the first half-space signal (SR1) has a directional characteristic which maximally attenuates a rear half-space (R2), and / or the second half-space signal (SR2) has a directional characteristic which maximally attenuates a front half-space (R1), wherein the front and rear half-spaces (R1, R2) are defined with respect to a frontal direction (F) of the carrier (T1).

5. Method according to one of claims 2 to 4, wherein a first auxiliary input signal (Eh1) is generated from the ambient sound (Us) by a first auxiliary input transducer (Mh1) of the first hearing instrument (H1), and / or a second auxiliary input signal (Eh2) is generated by a second auxiliary input transducer (Mh2) of the second hearing instrument (H2), and wherein the number (#N) of conversation partners (G1-3, Gj) of the wearer (T1) of the binaural hearing system (HS) is additionally determined on the basis of the first auxiliary input signal (Eh1) and / or the second auxiliary input signal (Eh2).

6. The method according to claim 5, wherein the first output signal (Out1) is generated based on the first input signal (E1) and the first auxiliary input signal (Eh1), and in particular based on the second input signal (E2) and / or the second auxiliary input signal (Eh2).

7. Method according to one of claims 2 to 6, wherein a speech component (SpA) in the ambient sound (Us) is monitored on the basis of the first and second input signals (E1, E2), wherein an angular direction (αj) of a sound source (SQj) of the ambient sound (Us) is determined on the basis of the first and second input signals (E1, E2), and wherein a presence and a position of an individual speaker are determined on the basis of the speech component (SPAj) and the angular direction (αj).

8. The method according to claim 7, wherein based on the first and second input signals (E1, E2) for said individual speaker - a length (Dj) of a conversation contribution (Bj) and / or - an overlap (ULj) of a conversation contribution (Bj) with a conversation contribution (BT) of the carrier (T1) is determined, and from this the individual speaker is determined as a conversation partner (G1-3, Gj) of the carrier (T1).

9. The method according to claim 7 or claim 8, wherein the presence and position of the individual speaker are determined only in one half-space (R1, R2) corresponding to one of the two half-space signals (SR1, SR2).

10. The method according to any one of claims 7 to 9, wherein a change in a position of the individual speaker is tracked.

11. Method according to one of the preceding claims, wherein when only one call partner (G1-3, Gj) is detected, a predetermined first value (V1) of an amplification parameter is applied to an intermediate signal generated on the basis of the first and / or second input signal (E1, E2), and -- exactly two call partners (G1-3, Gj) a second value (V2) of the amplification parameter is applied to the intermediate signal, which is 2 dB to 5 dB lower than the first value, and / or -- exactly three call partners (G1-3, Gj) a third value (V3) of the amplification parameter is applied to the intermediate signal, which is 4 dB to 10 dB lower than the first value.

12. The method according to claim 11 in conjunction with claim 4, wherein a gain and / or a noise gain and / or an offset of a background gain for the second half-space signal (SR2) is used as the gain parameter, which has a directional characteristic that maximally attenuates a front half-space (R1).

13. Hearing system (HS) with at least one first hearing instrument (H1), wherein the hearing system (HS) is configured to carry out the method according to one of the preceding claims.

14. Hearing system (HS) according to claim 13, designed as a binaural hearing system, which further comprises a second hearing instrument (H2) and which is configured to carry out the method according to one of claims 2 to 12

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