Method and system for improving the intelligibility of a group of conversing persons

Abstract

The invention relates to a method and a device for improving the intelligibility of a group of people conversing, each of whom can be both speaker and listener at times. The people are located at two or more positions P. A system (1) is used for this purpose, comprising two or more directional microphones M directed towards the positions P, two or more directional loudspeakers L, also directed towards the positions P, and a digital signal processor (DSP) capable of processing and forwarding acoustic signals in real time. Each microphone M continuously receives acoustic signals Ai and forwards them to the DSP. The DSP identifies a speech signal Si1 from each acoustic signal Ai in real time and generates a processed speech signal aSi1 from it. The DSP sends the processed speech signal aSi1 to one or more other loudspeakers L.The DSP recognizes subsequently arriving speech signals Si2, Si3 with the same characteristics as reflections of the speech signal Si1 and generates from them in real time processed reflected signals aSi2, aSi3, which it sends to another loudspeaker L within a specified time window (4) which lasts at most 40ms.

Description

[0001] The invention relates to a method for improving the intelligibility of a group of people conversing in an environment, preferably a room, in which each person in the group can be both speaker and listener at times, wherein the people are located at two or more stationary positions P1, P2, ... Pn. The invention also relates to a system for use in the said method. State of the art

[0002] In lectures in seminar rooms, community meetings, and similar events, there is typically one speaker and a large number of listeners. In this case, the speakers can use microphones, allowing loudspeakers to ensure good intelligibility for all listeners. Appropriate electronics can also eliminate background noise and generate further improvements in speech intelligibility.

[0003] In group conversations held indoors, for example in a restaurant hall, a garden restaurant, or around a conference table in an open-plan office, every speaker can also be a listener at times, and vice versa. Communication can be much more difficult in these situations. On the one hand, background noise, especially from other people present, is disruptive; on the other hand, reverberation can reduce the intelligibility of individual syllables.

[0004] To improve speech intelligibility in enclosed spaces, passive measures are used, such as sound-absorbing panels (e.g., as partitions), carpets, acoustic curtains, and similar items, which are intended to largely prevent resonance and echo formation. Depending on the building's structure, these can be difficult to install.

[0005] Active measures can include playing suitable noise or quiet background music as a sound masking system to reduce the disruptive effect of sound by covering it up. Such systems are also used to increase focus, concentration, and discretion in the workplace.

[0006] An additional difficulty arises during social group conversations, for example at a regular table in a restaurant, when someone wants to listen to a speaker who is sitting further away than directly next to or opposite them, and especially when a direct neighbor is having a separate conversation with another person in the group, speaking loudly and without wanting to hear it. For listeners with a hearing impairment, such a situation is even more challenging. Description of the invention

[0007] The object of the present invention is to present a method, as described above, that improves the intelligibility of other participants in a conversation. A further object is to improve the speech intelligibility of a selected speaker when several conversations are taking place simultaneously within the group. In addition, a system with which the presented methods can be carried out will be described.

[0008] The invention is solved by the features of the first claims of the respective categories. Improved variants are described in the dependent claims.

[0009] In the procedure described above, people are located at the aforementioned stationary positions P1, P2, ... Pn, resulting in a total of n positions. These can be in a room, an enclosed space such as a restaurant or office, or in a courtyard, a garden restaurant, or on a means of transport like a train, to name just a few examples. The positions P are essentially stationary during the execution of the procedure, although different positions can be chosen for the next arrangement. Typically, the positions are arranged around a table, as is usually the case in meetings and group discussions, but not in a train compartment.

[0010] According to the invention, a system is preferably arranged centrally with respect to the seats P. This system would, for example, be mounted in the middle of the table, preferably positioned above or below the heads of the participants, i.e., either standing on the table or hanging above it. It could thus be integrated into or concealed by a lamp, a table decoration, or the table itself. Generally, a suitable location is chosen that does not obstruct the participants' line of sight but maintains direct visual contact with all participants, so that the direct sound from all participants reaches the system directly and the impression of a direct conversation is preserved.

[0011] The system used for the method is described below. It comprises two or more directional microphones M1, M2, ... Mn, hereinafter referred to generally as microphones Mi, directed towards positions P1, P2, ... Pn, generally referred to as P; two or more directional loudspeakers L1, L2, ... Ln, hereinafter referred to as loudspeakers Lj, also directed towards positions P; and at least one digital signal processor (DSP) capable of processing and transmitting acoustic signals in real time and connected to all microphones Mi and loudspeakers Li (i, j=1... n) for signal transmission. Instead of a single DSP, several DSPs may be used, which are then connected to the loudspeakers and microphones accordingly. Since it is known to anyone skilled in the art that amplifiers must be present after the microphones and before the loudspeakers, these are not described further.

[0012] Ideally, one person is present at each seat P and remains relatively stationary. However, several people can also be in the area of ​​a single seat P if, for example, an extra chair is placed at the table. It is also possible for a person to move between two seats P.

[0013] The inventive method is characterized by the fact that Each microphone Mi continuously receives acoustic signals Ai and forwards them to the DSP; the DSP identifies a speech signal Si1 from location Pi in real time from each acoustic signal Ai of a microphone Mi and generates a processed speech signal aSi1 from it by separating it at least from the rest of the acoustic signal Ai and preferably also subjecting it to noise reduction (denoising / active noise reduction); and that the DSP sends the processed speech signal aSi1 to one or more other loudspeakers Lj, wherein the DSP recognizes subsequently arriving speech signals Si2, Si3, ..., which have the same characteristics as Si1, as reflections of the speech signal Si1 and generates similarly processed reflected signals aSi2, aSi3 from them in real time, which it sends to one or more other loudspeakers Lj within a predetermined time window (4) that lasts at most 40ms.

[0014] It has been found that syllable intelligibility or speech intelligibility is increased when the speech signal arrives several times in succession in slightly modified form. Naturally, the speech signal is reflected off various surfaces, walls, etc., and arrives at each listener, or rather at each seat P, with a time delay after the direct arrival of the speech signal. The system according to the invention additionally delivers the speech signal Si1 arriving at the system, as well as further reflections of it, as subsequently arriving speech signals Si2 and Si3, in a processed form to each listener, so that the human ear, or rather the brain, can perceive more intelligible syllables from the totality of the captured speech signals.

[0015] The processing of the initial and reflected speech signals Si1, Si2, Si3, ... into aSi1, aSi2, aSi3, ... is initially achieved by separating them from the remaining acoustic signal Ai. Acoustic filters can also be used. These can improve speech signals by reducing interference and unwanted noise, increasing clarity and intelligibility, and optimizing signal quality.

[0016] The most important techniques and principles are noise reduction, echo and reverberation suppression, frequency adjustment (equalization), removal of unwanted sounds such as clicks and pops, dynamic compression, beamforming, active noise cancellation (ANC), and speech enhancement through AI.

[0017] It should be noted that directional microphones Mi and directional loudspeakers Lj are used. A directional microphone Mi can be a combination of several microphones, a microphone with a passive reflector or screen, or a single omnidirectional microphone that is spatially selective due to algorithms. The directional microphones Mi thus primarily capture speech signals from the positions Pi to which they are assigned, while speech signals from other positions Pj are captured much less effectively. The same applies analogously to directional loudspeakers Lj: they primarily amplify the sound for the positions Pj toward which they are directed. People in adjacent positions to Pj barely hear the speech signals emitted by this loudspeaker Pj, or at least much less clearly.

[0018] In a preferred method, the processed speech signals aSi1, aSi2, aSi3, ... are transmitted to the loudspeakers Lj with a partial time delay, such that, measured from the arrival of the speech signal Si1 (t=0), they are all emitted within a time window of up to 10 and 50 ms, preferably between 10 and 30 ms, after t=0. Repetitions of the first speech signal arriving too early at a listener do not improve the sound quality, whereas those arriving within a time window of 10–30 ms afterward achieve the greatest effect. For this reason, the DSP waits to transmit the processed signals aSi1, aSi2, aSi3, ... to the loudspeakers Lj until 10ms after t=0 and refrains from transmitting the processed reflected signals if they could only be transmitted after 50ms, preferably only after 30ms after t=0, in order to avoid a reverberation effect.

[0019] The optimal time window depends on when the first direct sound reaches the listener and extends between 10 and 30 ms or 10 to a maximum of 50 ms thereafter. Since this exact time is unknown, the time of arrival of the first speech signal S1 at microphone M1 is approximated as t=0.

[0020] Preferably, the DSP has an initial routine that is performed at least at the beginning of use, and possibly also during a conversation, for example, periodically. This routine can determine the exact positions of the people located at positions P. If a position Pj is empty, the corresponding microphones Mj and loudspeaker Lj can be deactivated. The system can also automatically detect changes in the locations of people and consequently perform a repeated initial routine.

[0021] Based on the initial routine or other input, the time t=0 can be adjusted as needed, provided the distances of positions P to the DSP are known. This allows the time of arrival of the first direct sound from Pi at Pj to be estimated and compared with the detour the sound must travel from Pi via the DSP to Pj. A detour of approximately 34 cm corresponds to a time delay of 1 ms. Therefore, with a detour of approximately 1 m, the optimal time window begins 7 ms after the arrival of the first speech signal S1, i.e., 3 ms earlier. Given a signal processing time of approximately 2 ms by the DSP, the first signal in the example mentioned must be delayed by approximately 5 ms to achieve the desired total delay of 10 ms.

[0022] To further improve sound quality, the DSP can include feedback suppression, which prevents processed speech signals aSi1, aSi2, ... captured by microphones Mi, which were previously emitted by any loudspeaker Lj, from being reprocessed and sent to loudspeaker Lj.

[0023] The volume levels of the speakers Lj can preferably be adjusted individually, either manually or via a self-calibration function. Furthermore, it is possible to send the speech signals Si1 and their reflections aSi2, aSi3, ... in a processed form not only to the other speakers Lj (j#i) but also to Li. This allows the speaker to hear themselves better, thus reducing the need to speak louder, which would otherwise increase the noise level in the room, as can be observed in many restaurants.

[0024] In a further improved method, at most one person occupies each position P. Additionally, either several cameras Ki are positioned, each directed at a specific position Pi, or a single 360° camera K is used, which can selectively locate and analyze the individual positions Pi. Using either the cameras Ki or the camera K, it can be determined which speaker at position Pi a listener at position Pj is listening to. Consequently, only the processed speech signals aSi1, aSi2, ... from position Pi are transmitted to the loudspeaker at Pj. This makes it possible to converse with a person sitting far away without being disturbed by the conversations of other members of the group.

[0025] In a further improved version, it is even possible to select the language in which one wishes to hear a conversation. For this purpose, the DSP features a translation routine, allowing a listener at position Pj to select their preferred language. Subsequently, all processed speech signals aSi1, aSi2, ... are translated into this language before being sent to the loudspeaker Lj. The language selection can be made, for example, using an application software that can be run on a smartphone. Such software can also be used to adjust the desired volume and / or voice, ideally even individually for different loud or soft speakers at different positions Pj. The voice of the translated output can be modeled after the voice of the speaker.

[0026] In another preferred method, the DSP continuously improves its signal processing, assignments, volume settings, and / or calibrations using AI, for example, through machine or deep learning. This allows positions Pi and Pj to be adjusted over time as people move around, such as when additional people arrive. Several people can occupy a single position P, as long as no one-on-one conversations or translations are taking place.

[0027] The system according to the invention for carrying out a described process comprises two or more directional microphones (microphones) M1, M2, ...Mn, directed in different directions towards places P1, P2, ... Pn where people may be located during use, two or more directional loudspeakers (speakers) L1, L2, ... Ln, also directed towards places P1, P2, ... Pn, and at least one digital signal processor (DSP) capable of processing and transmitting acoustic signals in real time and connected to all microphones Mi and loudspeakers Lj for signal transmission, with i, j=1... n.

[0028] In particular, each directional microphone Mi can be a microphone array consisting of several compound microphones, a microphone with a passive reflector or screen, or a single omnidirectional microphone that is spatially selective due to algorithms.

[0029] Preferably, the system also includes one or more cameras K directed at seats Pi to detect which speaker at seat Pi a listener at seat Pj is currently facing.

[0030] Furthermore, the system can include a control interface with which at least the volume of the speakers Lj, and preferably also other settings such as the sensitivity of the microphones Mi and / or seats Pi of preferred speakers, can be adjusted. This control interface can preferably be configured via mobile phones or other wearable devices. A real-time translation routine with speech output, for translating processed speech signals aSi1, aSi2, ... into a preferred foreign language, can also be implemented. In addition, the speaker's voice can be imitated.

[0031] The system preferably comprises a housing in or on which the DSP, the microphones Mi, the loudspeakers Lj, and optionally one or more cameras K are mounted. The housing also includes a device for setting it up or hanging it. Brief description of the drawings

[0032] The invention is illustrated in the following drawings and explained in more detail with the aid of the reference numerals explained later. The drawings show: Fig. 1 a schematic view of two people at a table with the system according to the invention suspended above the table; Fig. 2 an acoustic time signal Ai as it is captured at the microphone Mi; Fig. 2 a processed speech signal aSi1 and further processed reflected speech signals aSi2, aSi3, ..., which are sent to a loudspeaker Lj; Fig. 3 an example of a table, viewed from above, with eight seats, with a system according to the invention arranged in the center. Ways to implement the invention

[0033] In Fig. 1 A room 2 with system 1 and two places Pi and Pj is shown to explain the procedure and the system 1 used for it.

[0034] The time signals in Fig. 2a and b These serve to explain, in particular the time delay of the output signals aSi1, aSi2, aSi3, .... Fig. 3 The figures depict the spatial relationships and distances of a typical group of, for example, eight people, as often arranged in restaurants or around meeting tables. These figures are not intended to be restrictive; they serve as illustrative examples.

[0035] The inventive system 1 and the inventive method will subsequently be explained in more detail with the help of all figures.

[0036] System 1 for carrying out a method according to the invention comprises two or more directional microphones Mi, also called microphones Mi, designated M1, M2, ...Mn. They are directed in different directions towards locations P1, P2, ...Pn where people may be present during use. Fig. 1 Two such places, Pi and Pj, are shown at a table with a reflective surface 3, where i, j = 1...n. In the example shown, the person at Pi is the speaker, and the person at Pj is the listener.

[0037] Each directional microphone (Mi) is typically either a microphone array consisting of several interconnected microphones, a microphone with a passive reflector or screen, or a single omnidirectional microphone that is spatially selective due to algorithms. An amplifier is usually located after each microphone (Mi).

[0038] System 1 also includes two or more directional loudspeakers Lj, also called loudspeakers Lj, designated L1, L2, ... Ln. These are also directed towards positions P1, P2, ... Pn, and typically an amplifier is positioned in front of each loudspeaker Lj. At least one digital signal processor (DSP) of System 1, capable of processing and transmitting acoustic signals in real time, is connected to all microphones M and loudspeakers L for signal transmission. If System 1 includes multiple DSPs, these are connected to each other as well as to the microphones M and loudspeakers L.

[0039] In a preferred version, the system 1 may also include cameras K1, K2, ... Kn, which are directed at seats P1, P2, ... Pn, or a single camera K with a 360° detection angle, which can identify the individual seats P.

[0040] In Fig. 3A table with a reflective surface 3 is shown, on which eight seats P1, ... P8 are provided. System 1, comprising microphones Mi, loudspeakers Lj, and optional cameras K, is positioned centrally on or above the table, for example, in combination with table decorations or a lamp. The system is preferably positioned centrally with respect to seats P so that it can receive direct sound waves from the speakers and thus ensure that the path lengths of all speech signals from the speakers to System 1 are as short and not too dissimilar as possible.

[0041] Instead of a real person, a virtual person can also participate in the conversation at a location P, for example with a monitor equipped with a microphone and speakers. The voice transmission can then be either acoustic via the speakers and microphone on the monitor or via cable directly to the DSP.

[0042] System 1 preferably comprises a housing 1 in or on which the DSP, the microphones Mi, the loudspeakers Lj and optionally one or more cameras K are mounted. It includes a device for setting up or hanging the housing.

[0043] In Fig. 1 The direct sound path from Pi to Pj is shown with dashed arrows, as well as two further sound paths that, originating from the speaking person at Pi, are reflected on the reflective surfaces 3 of the table or the lower edge of the system 1 and reach the person at Pj. These sound paths occur naturally and are not influenced by the inventive method.

[0044] System 1 can, in particular, include a control interface 5 with which at least the volume of the loudspeakers Lj, and preferably also further settings such as the sensitivity of the microphones Mi and / or the positions P of preferred speakers, can be set. The control interface 5 can preferably be set via mobile phones or other wearables of the users, or one or more modules 5 similar to a remote control are provided. In addition, System 1 can include a real-time translation routine with speech output for translating processed speech signals aSi1, aSi2, ... into a preferred foreign language. The preferred language would preferably be adjustable via a control interface 5.

[0045] The method according to the invention is described below. It serves to improve the intelligibility of a group of people conversing in an environment, preferably a room 2, where each person can be both speaker and listener at times, and where the people are located at two or more places P1, P2, ... Pn. These places P remain stationary within a predetermined area during the group conversation. Seats at chairs around a table can be moved, but seats at adjacent tables are not included. The method is carried out using a system 1 described above, which is preferably arranged centrally with respect to the places P.

[0046] Each microphone Mi continuously receives acoustic signals Ai and forwards them to the DSP. Fig. 2aA continuous signal Ai, as captured by microphone Mi, is shown. It consists mostly of background noise H. As soon as the person speaks at Pi, an initial signal Si1 is captured by microphone Mi because this person is within the detection range of the directional microphone Mi. Each syllable spoken is also reflected as a sound wave at the reflective surfaces 3 and later reaches microphone Mi, which is identified by the DSP as Si2, Si3, ...

[0047] The DSP identifies a speech signal Si1 from location Pi from each acoustic signal Ai of a microphone Mi and generates a processed speech signal aSi1 from it in real time by separating it from the rest of the acoustic signal Ai and preferably also subjecting it to active noise reduction. Fig. 2bThe processed speech signal aSi1 is shown. The DSP sends the processed speech signal aSi1 to one or more other loudspeakers Lj. Since the processing of each signal through the DSP takes a certain amount of time, usually about 2-4 ms, the processed speech signal aSi1 is sent to loudspeaker Lj later than the speech signal Si1 is detected.

[0048] All subsequently arriving speech signals Si2, Si3, ... which have the same characteristics as Si1, are recognized by the DSP as reflections of the speech signal Si1. The DSP then generates these reflected signals aSi2, aSi3, which are processed in real time and sent to one or more other loudspeakers Lj within a predefined time window 4, lasting a maximum of 40 ms.

[0049] In Fig. 2bThe processed reflected speech signals aSi2 and aSi3 are shown with a time delay following the processed speech signal aSi1. All of them are amplified and sent from loudspeaker Lj to position Pj. Reflected speech signals aSi that are processed too late, i.e., available only after time window 4 has elapsed, are discarded.

[0050] In Fig. 1 These speech signals emitted by the loudspeaker are shown as aSi1, aSi2, and aSi3. They reach the person at Pj in addition to the direct sound and the speech signals reflected from it naturally; they are shown as dashed lines and are not further labeled.

[0051] The person at Pj now hears and understands the syllables from the person at Pi much more clearly. This is partly because more information about these syllables reaches the system 1, and partly because their volume has been amplified.

[0052] It has been shown that the processed speech signals aSi1 and the reflected processed speech signals aSi2, aSi3 contribute more to syllable intelligibility when they are partially delayed or not transmitted to the loudspeakers Lj at all. Measured from the arrival of the speech signal Si1 at the DSP (t=0), they should preferably all be transmitted within a time window of 10 to 50 ms, preferably between 10 and 30 ms, after t=0. Speech signals Si2, Si3, ... arriving too late are therefore no longer sent to the loudspeakers.

[0053] Otherwise, a reverberation effect or unwanted sound coloration could occur. Therefore, time window 4 lasts a maximum of 40 ms, preferably 30 ms or 20 ms.

[0054] In Fig. 2bAs indicated by the dashed arrow, the processed speech signal aSi1 was transmitted with a time delay to avoid arriving at listener Pj too early. The processed speech signal aSi3, on the other hand, would be transmitted more than 30 ms after t=0, which could create an echo effect. Therefore, in a preferred version, it is not transmitted if the time window, in Fig. 2b labelled 4', was set to 20ms.

[0055] The best results are achieved with a time window of 4 of 20ms, which begins 10ms after the direct sound arrives at the listener and ends 30ms later.

[0056] Preferably, the DSP includes feedback suppression, which prevents processed speech signals aS captured by the microphone Mi, which were previously emitted by any loudspeaker Lj, from being reprocessed and sent to one or more loudspeakers Lj.

[0057] The DSP preferably has an initialization routine that is executed at the start of use. The volume levels of the loudspeakers Lj can preferably be adjusted individually, in particular manually or by means of a self-calibration function.

[0058] In a preferred method, at most one person occupies each seat P. Using one or more cameras K directed at seats P, it can be determined which speaker at seat Pi a listener at seat Pj is listening to. Consequently, only the processed speech signals aSi1, aSi2, ... originating from seat Pi are routed from the DSP to the loudspeaker Pj. This method also allows people sitting far apart to converse.

[0059] In Fig. 3The person at seat P1 can therefore easily communicate with someone sitting at P4 or P5, which would otherwise be nearly impossible in a crowded restaurant. Even conversations from seat P1 with people at P3 or P6 are often difficult if other lively conversations are taking place at the table.

[0060] At conferences, but also in restaurants, it often happens that only one person speaks at a time, while the others listen and contribute to the conversation in turn. The acoustics, and thus the quality of the conversation, are significantly improved in this case as well by the inventive method. One consequence of this is that people can speak more quietly, which further improves the atmosphere of the conversation.

[0061] Furthermore, System 1 can include a translation routine, allowing a listener at position Pj to select their preferred language. All processed speech signals aSi1, aSi2, ... are then translated into this language before being sent to loudspeaker Lj. Since Lj are directional loudspeakers, Pj's neighbors are not disturbed. It is also possible to send the processed speech signals aSi1, aSi2, ... to headphones, but due to the time delay, this is only practical for translations or if the listener uses a hearing aid. Combining the processed signals with natural, direct sound is then prevented or at least severely limited.

[0062] The DSP can continuously improve its signal processing, mappings, volume settings, and / or calibrations using machine learning. This allows, especially when cameras are used, the time delay to be adjusted based on the distances between people. Speaker outputs to P8 in Fig. 3 Signals originating from P1 therefore require less time delay than those originating from P4, since the direct sound from P4 to P8 takes approximately the same amount of time to travel as a speech signal via system 1. The direct sound from P1 to P8, however, arrives at P8 significantly earlier than the signal traveling via system 1. Reference symbol list

[0063] PPlatz, Plätze P1, P2, ...Platz 1, Platz 2, ... MiRichtmikrofon or directional microphones, also microphone resp. microphones, i=1... n M1, M2, ...microphone 1, microphone 2, ... Lj directional loudspeaker, also called loudspeaker, j=1 ... n L1, L2, ...loudspeaker 1, loudspeaker 2 K camera or cameras K1, K2, ...camera 1, camera 2, ... DSP digital signal processor(s) A acoustic signals at microphone i captured Si speech signal from the acoustic signal at microphone Mi captured H background noise aS processed speech signal, generally aSi1 first captured processed speech signal from the acoustic signal at microphone Mi captured, separated from background noise Si2, Si3, ... later arriving speech signals from Si1, reflected speech signals from Si1 aSi2, aSi3 processed reflected speech signals from Si1, from the acoustic signal at microphone Mi captured, separated from background noise i, j counter, i=1... n; j=1...n 1System, housing 2Room 3Reflection surface 4Time window, general, 4' time window with a duration of 20ms 5Control interface.

Claims

1. Method for improving the intelligibility of a group of people conversing in an environment, preferably a room (2), in which each person of the group can be a speaker at times and a listener at other times, wherein the persons are located at two or more stationary positions P1, P2, ... Pn, using a system (1) preferably arranged centrally with respect to the positions P, comprising: - two or more directional microphones (microphones) M1, M2, ... Mn directed towards the positions P1, P2, ... Pn, - two or more directional loudspeakers (loudspeakers) L1, L2, ... Ln, also directed towards the positions P1, P2, ... Pn, - at least one digital signal processor (DSP) capable of processing and transmitting acoustic signals in real time and connected to all microphones Mi and loudspeakers Lj for signal transmission (i, j=1... n), characterized by the fact that- each microphone Mi continuously receives acoustic signals Ai and forwards them to the DSP; - the DSP identifies a speech signal Si1 from location Pi in real time from each acoustic signal Ai of a microphone Mi and generates a processed speech signal aSi1 from it by separating it at least from the rest of the acoustic signal Ai and preferably also subjecting it to active noise reduction, - and that the DSP sends the processed speech signal aSi1 to one or more other loudspeakers Lj, - wherein the DSP recognizes subsequently arriving speech signals Si2, Si3, ... which have the same characteristics as Si1, as reflections of the speech signal Si1 and generates similarly processed reflected signals aSi2, aSi3 from them in real time, which it sends to one or more other loudspeakers Lj within a predetermined time window (4) that lasts at most 40ms.

2. Method according to claim 1, characterized by the fact thatEach directional microphone Mi is a microphone array consisting of several compound microphones, a microphone with a passive reflector or screen, or a single omnidirectional microphone which is spatially selective due to algorithms.

3. Method according to claim 1 or 2, characterized by the fact that the processed speech signals aSi1 and the processed reflected speech signals aSi2, aSi3, ... are transmitted to the loudspeakers Lj with a partial time delay, so that they are all emitted in a time window (4) between 10 and 50ms, preferably between 10 and 30ms after t=0, where t=0 is defined by the arrival of the speech signal Si1 at the DSP.

4. Method according to any one of the preceding claims, characterized by the fact thatThe DSP includes feedback suppression, which prevents processed speech signals aSi1, aSi2, ..., captured by microphones Mi and previously emitted by any loudspeaker Lj, from being reprocessed and sent to loudspeaker L.

5. Method according to any one of the preceding claims, characterized by the fact that The DSP has an initial routine and performs it at least at the beginning of use.

6. Method according to any one of the preceding claims, characterized by the fact that the volume levels of the speakers Lj and / or the time windows (4) of each microphone Mi to each speaker Lj can be individually set, in particular manually or by means of a self-calibration function.

7. Method according to any of the preceding claims, characterized by the fact thatthat at most one person is at each seat and that by means of one or more cameras K, which are directed at seats P, it can be determined which speaker at seat Pi a listener at seat Pj is listening to, so that as a result only the processed speech signals aSi1, aSi2, ... from the DSP to the loudspeaker at Pj are directed.

8. Method according to claim 7, characterized by the fact that The DSP has a translation routine whereby a listener at seat Pj can select their preferred language and then all processed speech signals aSi1, aSi2, ... are translated into this language before being sent to the loudspeaker Lj, preferably also imitating the speaker's voice during speech output.

9. Method according to any one of the preceding claims, characterized by the fact thatThe DSP continuously improves its signal processing, mappings, voice imitations, volume settings, individual time windows (4) and / or calibrations, preferably through machine and / or deep learning, using AI.

10. System (1) for carrying out a method according to one of the preceding claims, comprising: - two or more directional microphones (microphones) M1, M2, ...Mn, which are directed in different directions towards places P1, P2, ... Pn where persons may be located during use; - two or more directional loudspeakers (loudspeakers) L1, L2, ... Ln, which are also directed towards places P1, P2, ... Pn; - and at least one digital signal processor (DSP) which can process and transmit acoustic signals in real time and is connected to all microphones Mi and loudspeakers Lj for signal transmission (i, j=1...n).

11. System according to claim 10, characterized by the fact thatEach directional microphone Mi is a microphone array consisting of several compound microphones, a microphone with a passive reflector or screen, or a single omnidirectional microphone Mi which is spatially selective due to algorithms.

12. System according to one of claims 10 or 11, characterized by one or more cameras Ki, directed at seats Pi, to record which speaker at seat Pi a listener at seat Pj is currently facing.

13. System according to one of claims 10 to 12, characterized by a control interface (5) with which at least the volume of the loudspeakers L, preferably also other settings such as the sensitivities of the microphones M and / or seats Pi of preferred speakers can be set, wherein the control interface (5) can preferably be set via mobile phones or other wearables of the users.

14. System according to claim 13, characterized bya real-time translation routine with speech output for translating processed speech signals aSi1, aSi2, ... into a preferred foreign language, wherein the speech output can preferably also imitate the voice of the speaker.

15. System according to one of claims 10 to 14, comprising a housing (1) in or on which the DSP, the microphones Mi, the loudspeakers Lj and optionally the one or more cameras K are attached, wherein the housing (1) comprises a device for setting up or hanging.

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