WO2016020511A2

WO2016020511A2 - Method for reducing the comprehensibility of speech signals, and separating component for influencing sound transmission

Info

Publication number: WO2016020511A2
Application number: PCT/EP2015/068222
Authority: WO
Inventors: Philip Leistner; Horst Drotleff; Andreas Liebl; Xiaoru Zhou
Original assignee: Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e. V.
Priority date: 2014-08-08
Filing date: 2015-08-07
Publication date: 2016-02-11
Also published as: EP3178184A2; DE102014111365A1; EP3178184B1; WO2016020511A3

Abstract

The invention relates to a method for reducing the comprehensibility of a speech signal, involving the following steps: a) detecting the amplitude, variation over time, and frequency of the speech signal (8); b) generating a masking sound (9) which is dependent on the amplitude, variation over time, and frequency of the speech signal (8) in such a way that an incomprehensible superimposed sound (12) is created by superimposing the masking sound (9) on the speech signal (8), said superimposed sound (12) not being significantly louder than the speech signal, the masking sound (9) being generated in a randomly changing manner, and the speech signal (8) being split into intervals, characterized in that the length of the intervals is randomly modified according to a randomization function. The invention also relates to a separating component for separating a room (2) from a room environment (3) using sensors (4), actuators (5) and a signal processing unit (6) in order to influence sound transmission, characterized in that the separating component (1) includes the following parts: sensors (4) for detecting the speech signals (8) from the room (2); actuators (5) for generating a masking sound that is dependent on the amplitude, variation over time, and frequency of the speech signal (8); the sensors (4) and the actuators (5) being located at a distance from one another so that the speech signal (8) has a certain propagation time until the masking sound (9) is superimposed thereupon in order to take into account propagation times of the speech signal (8) in the separating component (1).

Description

Patent application: Method for reducing the intelligibility of speech signals and separating component for influencing the sound transmission

Technical area

The invention relates to a method for reducing the intelligibility of speech signals and a separating component for influencing the sound transmission. It should serve to avoid disturbances and fend off eavesdropping attacks. State of the art

There are two main motivations to reduce the intelligibility of a speech signal. It is known that speech signals distract a person more than signals with no apparent information content, even if the volume is higher. This applies to some extent even if the signals have no apparent

Information content louder than the voice signal. Distraction by voice signals occurs in open-plan offices, for example. Precisely because of sound insulation and

Sound attenuation a low volume level can be achieved in open-plan offices, the distraction is relevant by a voice signal. One approach is to superimpose the speech signal with a noise signal, making it incomprehensible

Overlay sound is created. The disadvantage of this is that the overlay sound has a comparatively high noise level.

The intelligibility of speech signals should often be lowered to make it difficult to eavesdrop. In the wiretapping security of conversations, the telecommunications-technical eavesdropping on radio or cable transmissions is undoubtedly in the foreground. Technologically speaking, these information technology can be

Protect systems by means of special technical measures and behavior, too if it temporarily leads to limited efficiency and convenience. The situation is different with the spoken word in rooms of buildings, even if Faraday cages are present, data lines are cut off and mobile communication devices are switched off. Even through thick concrete walls can be achieved with simple technical aids, eg. As structure-borne sound sensors, the word spoken in space and understand understandable. In the building envelope - whether opaque or transparent - modified laser vibrometers allow the invisible and unnoticed

Eavesdropping even from a distance. In "normal buildings", sensitive microphones or simply listening in otherwise quiet surroundings suffice.After all, the various building services installations, eg heating pipes or ventilation ducts, are sound transmitters throughout the building.Constructional and technical recommendations according to the state of the art for minimizing the For this reason, interceptibility includes, for example, the omission of windows, a complete solid construction, protection against structure-borne sound propagation and absolutely unadorned spaces

However, they completely contradict the ideas of the users.

A viable solution must therefore be found in typical building structures and structures

Separating components, for. As facades, walls, ceilings, etc., can be integrated to achieve the most complete secure listening rooms. This applies both to the new building as well as for the retrofitting of existing rooms. Starting points are the structural-technical substance and the contemporary components of office buildings and their available sound insulation potential. Since these are mostly multi-layered and multi-layered facades, walls, ceilings, etc., their realistic sound insulation values are between approx. 25 and 55 dB, and for heavy floor slabs a little above that. These values can of course be increased, but only by more mass or more expensive

Construction technologies. These expensive and complex technical approaches to increase the sound insulation also include active systems, consisting of (vibro-) acoustic sensors, actuators and a signal processing, to influence the

Vibration Behavior of Separation Components ("Active Control of Vibration" Elsevier Ltd. 1996, "Active control of noise and vibration" RC Press 1 996), e.g. Plates or discs (DE 1 02005044448, DE 1 98261 71). The influence can be as

Vibration damping or superposition done. It proves to be particularly valuable at low frequencies, because in this area the sound insulation of all walls, Ceiling etc. total or partially resonance-like low. However, the frequency range of intelligible speech is at medium and high frequencies, where an increased sound attenuation with active systems is not or hardly achievable.

In contrast, one approach to reducing the ability to intercept is to overlay speech with masking signals so that a sound may be heard or detected but its content is incomprehensible. On the one hand, these masking systems must alienate the outgoing voice content so that it can not be decrypted. Both the word and the sentence intelligibility must therefore be reduced to a semantically unconstructable measure. On the other hand, no noticeable acoustic disturbance (annoyance) may occur in the room or outside, since this depends on the acceptance by the users. An all-drowning noise is therefore not an acceptable solution as it would have to significantly exceed the speech signal level. Conventional systems for the confusion of confidential

Language information, such. The device described in "http://www.global-security.org/de/produkte/lauschabwehr/76-akustisches-verrauschungssystem" is therefore subject to severe limitations in terms of its practical application In the case of electronic methods, the coherent speech signal is at least partially extracted from the incoherent noise A method is known from US 3,213, 199 A in which sound is recorded in a room with a microphone

Neighboring room separately. To prevent in-room conversations in the room

Neighboring room is bugged, a speaker is provided, which depends on the frequency and the amplitude of the sound recorded in the room at any time

Neighborhood generates a masking sound.

From the CA 2,461,148 a special method of generating masking sound using a frequency filter is known. The sound to be masked or encoded is decomposed into time blocks. The frequency spectrum and the power level are determined and it is continuously broadband sound with a matching

Generated level spectrum for masking. Here is the amplitude of the masking sound relatively small when it comes to pure masking applications, and high when encryption is to be achieved.

From US 201 2/031 689 A1 is an electronic device for the production of

Masking sound with a variety of microphones and a speaker known. Speech signals and ambient signals are recorded. Depending on this, a masking sound is generated.

From DE 1 0 2010 014 819 A1 a structure for an active noise suppression in a vehicle interior is known. The active noise suppression takes place depending on the level of the sound to be suppressed. It is taken into account that after damping the sound to be suppressed the active noise suppression can be weaker.

Also WO2004 / 107806 A1 discloses a system with which sound can be masked.

US 2005/0065778 A1 discloses an apparatus and a method for masking speech sound. The amplitude and the frequency of the

Masking signal can be varied in certain periods of time. The time periods may have a fixed, variable or random length. Using a so-called "babble generator", a speech-like signal, an artificial voice is generated as a masking signal, and then the artificial speech signal generated in the "babble generator" is superimposed on the masking signal with the actual speech signal.

The demands on largely tap-proof separation components with attached

Masking systems for a room thus consist both in minimizing speech intelligibility and preventing reconstructability of the outwardly-urging speech signal as well as in a minimized acoustic disturbance potential for the users of the adjoining rooms.

The object of the invention is therefore to provide a method with which a speech signal can be made incomprehensible in an improved manner. It is also a task Improved privacy from rooms where confidential discussions are conducted. The solution is given in the independent claims. The dependent claims teach advantageous embodiments. The description and the drawings give further explanation.

Walkthrough

It has been recognized that, for this purpose, a method for reducing the intelligibility of a speech signal with the following steps is to be provided.

a) detection of amplitude, time and frequency of the speech signal; where time as a parameter is not the time of signal acquisition but

Stickler, ES .; Bigenzahl, W .: Vocal diagnostics. Springer-Verlag Wien, 201 3 (from page 1 53) known and accepted by those skilled in the temporal fluctuation of the language, that is the temporal change of the frequency amplitude spectrum.

b) generating a masking sound, the amplitude, time and frequency of the

Speech signal is so dependent that arises by superimposition of the speech signal with the masking sound an incomprehensible overlay sound, wherein the

Overlay sound does not have a much higher volume than the voice signal.

The core idea is therefore that the masking sound depends in a differentiated manner on the speech signal whose intelligibility it is to be lowered, and is specifically adapted to the speech signal. The prior art is that a masking sound with a higher

Volume is provided when the speech signal is louder. On the other hand, the invention uses a masking sound which is matched to the speech signal and, in addition to the amplitude, also takes account of the time and frequency of the speech signal, can already be superposed with a volume in the range of the volume of the speech signal to form an incomprehensible overlay sound. So it is possible that the

Overlay sound does not have a much higher volume than the voice signal. The masking sound itself can be a more or less understandable speech sound. In the prior art is due to non-specific noise, which louder than that

Speech signal is, the speech signal simply drowned out, so that the overlay sound is not understandable. In the present case, the masking sound could also be quieter than the voice signal. Of course, the overlay sound is a bit louder than the voice signal, but not significantly louder.

It should be noted that the generation of the masking sound must be done in real time, since for the superposition of the masking sound must be present simultaneously with the speech signal. Particularly complicated, computationally expensive process for the production of

Masking sounds are therefore eliminated. As shown again by an example below, the detection of the speech signal, the generation of the masking sound and the consequent superimposition take place at different locations. Thus, the speech signal has a certain duration, until it comes to the superposition with the masking sound. This time can be used to create the masking sound.

In the present case, the speech signal is split into intervals. The masking sound is formed by temporal inversion of the short intervals. To put it clearly:

While the speech signal contains forward spoken text in the usual way, the masking sound is, as it were, "spoken" sequences backwards. The overlay sound thus results from the superposition of the speech signal with temporally inverted sequences of the signal. Already due to the above

In order to provide the masking sound in real time, no long sequences can be inverted. Rather, these are short intervals, for example, have a length of 20 ms to 200 ms, in particular 50 ms to 100 ms, d. H. the length of the intervals is arbitrarily variable in these interval limits.

It should be mentioned at this point for the understanding that the masking sound is formed directly from the speech signal. That is, according to the invention, only the

Speech signal used and changed to then superimpose the speech signal in a modified form as a masking sound with itself. In contrast, in US Pat. No. 3,213,199 A, a new signal is generated from the amplitude and the frequency of the primary signal, that is to say of the detected speech signal.

Accordingly, the generation of the masking sound is subject to a random change. It is understood that this random change must remain within a framework that ensures that the overlay sound remains incomprehensible. The lowering of the intelligibility of a speech signal can also serve the eavesdropping of a Prevent conversation. In this case, a generation of the masking sound according to an always unchanged method, according to the invention, the change in the masking results not from the change in the masking rule but from the variation of the interval length, cause the process is detected by the eavesdropper and thus a reconstruction of the speech signal would be possible. By the random

Changing the generation of the masking sound makes such a reconstruction considerably more difficult. The way to random change is to change the length of the short time intervals described above according to a random function. It should be noted that the method described in US 2005/0065778 A1 is different. In US 2005/0065778 A1, the manner in which a masking sound is generated within the time intervals as a function of the detected speech signal is to be changed. This variation should not be ruled out in the present case.

What is decisive, however, is that in the present case a random change of the time intervals has been recognized as sufficient to increase the overall masking requirement

encrypt.

Based on the above considerations, a variety of methods are conceivable with which the masking sound can be generated.

In one embodiment, the time intervals are changed randomly in a range of 20 ms to 200 ms, in particular in a range of 50 ms to 100 ms. In one embodiment, an insulation and / or attenuation of the speech signal is taken into account in the generation of the masking sound, wherein in particular a

Masking sound is provided at low volume, when the overlay sound from the superposition of the masking sound with the insulated and / or attenuated speech signal. It will be possible in many cases to insulate or dampen the speech signal before it enters an area in which a disturbance or a bugging attack is to be fought off. Thus, a masking sound with lower volume is sufficient. Accordingly, the overlay sound is also less loud, so that possible disturbances are further lowered. As far as above a comparison between the volume of the masking sound, the overlay sound and the

Speech signal is made, it is useful to turn off the speech signal after an eventual insulation or damping. As mentioned above, it is particularly important to increase the privacy of rooms. In addition to the described method, the invention therefore also relates to a separating component for separating a room from a room environment. In the

Room environment is usually the exterior of a building. But it is also conceivable that it is a neighboring room in a building. The separating component contains sensors, actuators and a signal processing unit for influencing the sound transmission. The separating component is characterized by the following components:

There are sensors for detecting the voice signal coming out of the room. Furthermore, there are actuators for generating a masking sound. The masking sound depends on the amplitude, time and frequency of the speech signal. As already explained above, it is not enough to the volume alone

take into account, but also time and frequency of the speech signal is too

consider. It is essential for this purpose that a signal processing unit is provided which can process the signals detected by the sensors so that the

Actuators can generate such a masking that an incomprehensible overlay sound is created by superimposing the masking sound with the speech signal in the room environment, without the overlay sound must have a much higher volume than the speech signal. The masking sound itself has about the volume of the speech signal in the room environment, but can also be quieter.

The sensors and the actuators are spaced in a defined manner, so that the speech signal has a certain transit time until it comes to superimposing with the masking sound. In US 3,213, 1 99 A, the sensors and actuators are also spaced. As has been recognized in the context of the present invention, the distance can not be considered or even used. In a microphone as a sensor in the middle of the room, as shown in FIG. 1 of US Pat. No. 3,211,399 A, the speech signal has arrived faster from a speaker near the dividing wall at the dividing wall than at the microphone. For a speaker sitting away from the partition, the voice signal is more likely to be on the microphone than on the partition. A fair

Consideration of the transit time of the speech signal from the sensors to the actuators is thus not possible in US Pat. No. 3,221,399A. This is for the execution of the in US No. 3,213,199 A also not required, because the masking sound is not generated and superimposed in a timely manner using the primary speech signal itself, but only from its specific parameters frequency and amplitude. In the present separation component, however, this is possible as described below and also required to fulfill the task according to the invention.

Accordingly, the signal processing unit is designed to delay the masking sound with respect to the speech signal in time to run times of the speech signal in

Separating component to be considered. It should be noted that the duration of the electrical signal from the sensors and the signal processing unit to the actuators is usually much shorter than the duration of the sound in the separation component. If the sensors in the area of the side of the separating component facing the room and the actuators in the region of the side of the separating component facing the room environment are now to be taken into account, this duration of the sound from the sensors to the actuators must be considered. This transit time is advantageously used according to the invention in that the time delay resulting from the signal propagation time is sufficiently designed such that the signal processing unit transmits the masking sound directly, i. generate in real time and provide for the overlay.

In one embodiment, the separation component is designed so that the generation of the masking sound is subject to a random change. Thus, the generation of the masking sound can be reproduced, if at all, only with great difficulty. This prevents the speech signal from being reconstructed. As already shown in the description of the procedure, the change must be within limits that allow the generation of incomprehensible overlay sound.

In an important embodiment, the separating component is designed so that the method described above can be carried out with all its advantages.

In one embodiment, the separating component is designed to be multi-layered or multi-shelled. Such structures are common and readily available. In addition, gaps are realized, so that there about the signal processing unit can be accommodated well. In one embodiment, the separating component is set up in such a way that the masking sound is superposed with the speech signal within the separating component. It can thus be achieved that in a region of the room environment facing the room environment

Separating component forms a structure-borne sound that radiates incomprehensible overlay sound into the room environment.

Alternatively, under certain conditions, in addition, the separating component is set up so that in the room environment, a superposition of the passing through the separating member portion of the speech signal with the masking sound. This is especially the case when the actuators for generating the masking sound are located on the side of the separating component facing the room environment. In this case, a portion of the speech signal greatly reduced by the separating component passes through the

Separating member. This is then, as it were only in the room environment by an overlay with the masking sound to an incomprehensible overlay sound.

Both approaches have advantages. A corresponding overlay within the

Separating component, so that as in the first described alternative in which the

Room environment facing region of the separation component forms a structure-borne sound that radiates incomprehensible overlay sound into the room environment, a wiretapping attack should be even more difficult. A tapping the structure-borne sound on the room environment facing side is no longer possible. In one embodiment, the sensors are designed as vibroacoustic sensors. Vibroacoustic sensors are suitable for the accurate detection of a speech signal.

In one embodiment of the invention, the sensors for detecting the

Speech signal attached to the room-facing side of the separating member or in this area. There, the speech signal has the highest intensity, so that sensors there can capture the speech signal well. Alternatively or additionally, the actuators for generating the masking sound on the room environment facing side of the separating member or in this area are attached. The masking sound is in the

Room environment required, so that attachment of the actuators in this area makes sense. Of course, it is also conceivable to attach the actuators to a position near the room, for example in the case of a multi-layered separating component. It is understood that all the above measures can not make a eavesdropping attack completely impossible. However, the invention can make an important contribution to complicate such eavesdropping significantly. This is very desirable in many cases. embodiment

FIG. 1, which shows a schematic representation of a double-shell partition wall with a vibroacoustic system integrated according to the invention for masking the speech sound transmission between two rooms, will be described below

Invention will be described in more detail. Starting point is, as can be seen in Figure 1, a multi-layered separating member 1, z. B. a double-shell wall or ceiling construction or a single-shell wall with front shell, with adequate sound insulation in the information-containing

Voice frequency range. This acoustic transmission path between the

listening protected room 2 and its external environment forming the room environment

Environment 3 is non-linearly manipulated by means of a vibroacoustic speech signal masking system. It comprises (vibro-) acoustic sensors 4 and actuators 5 and a signal processing module 6, which are integrated into the multi-shell separating component 1. In this speech masking method, a modified speech signal 9 serving as a masking sound is generated in real time from the primary speech signal 8 detected on the room side 7 of the separating component 1, which is fed by means of the actuators 5 into the multi-layered separating component 1 or, preferably, its outer side 1 0 Sound emission stimulates. The type, number and distribution of the sensors 4, actuators 5, etc. depends, inter alia, on the construction of the separating component 1. Finally, the superimposed radiation of both signals, ie the insulated primary sound signal 1 1 and the modified voice or masking signal 9, on the outside 10 leads to a quiet and incomprehensible overlay sound 1 2. The above-mentioned generation of the modified speech signal 9 in Real time should simply mean that the superposition of the primary speech signal 8 and the modified speech signal 9 to the overlay sound in the external environment 3 must be done simultaneously. The primary speech signal 8 needs from the vibrating acoustic sensors 4 to the Actuators 5 a certain duration. This is advantageous in that the generation of the modified speech signal 9 takes some time.

This system can also be used in conjunction with solid components. Here is z. As the possibility, the sensors 4 and 5 actuators, similar to

Junction boxes for electrical power or data networks, to be integrated directly into the component. Alternatively, the component room side can also be provided with a header shell, for ceilings z. B. also in the form of a floating screed, the

Voice masking system finds its place in the gap between header shell and solid component. The attachment shell generally does not serve sound-insulating purposes, but can also act as a sound absorber in room 2 to improve room acoustics. In any case, the Mehrschaligkeit or multi-layer offers the advantage that the emitted from the outside 10 of the separating component 1 masking sound 9, the space 2 is not or only weakly achieved space and thus is not perceived as disturbing. In this aspect, an asymmetrical structure of the separating component 1 is of particular importance.

The algorithm for generating the modified speech signal 9 accesses the detected primary speech signal 8. The detection is followed by filtering, time delay and temporal inversion of separate signal sections. The known sound transmission properties (level, phase and

Frequency behavior) of the separating component 1 taken into account. As a result, only a kind of "Babylonian linguistic confusion" of a low level can be perceived on the outside 10 of the separating component 1 instead of a quieter speech, making the speech and speaker recognition and the associated inaudibility considerably more difficult or even completely prevented.

The filtering, time delay and temporal inversion of short signal sections is based on the recognition, see, for example, US Pat. B. Sound Masking Performance of Time-Reversed Masker Processed from the Target Speech. Acta Acustica United with Acustica. Vol. 98 (2012), pp. 1, 35-141, that this manipulation, with simultaneous presentation of the

Original speech signal achieves a significantly higher masking effect, as the superposition with the known, usually differently filtered noise signals. This realization was created in the context of language masks between workplaces in open-plan offices in order to reduce the disruptive effect of intelligible main language.

Abhörsicherheit or the like is there no relevant goal. The speech masking according to the invention depends on the language prevailing in the application, since the speech patterns differ. Although nothing changes at the basic manipulation steps, some details, e.g.

Filter corner frequencies and algorithms can be optimized language-specifically. As an embodiment, therefore, the language-dependent selection and adaptation of

System features provided.

In this context, an algorithmic additional module for

Speech recognition can be integrated in two main ways. On the one hand, it distinguishes speech from other non-speech sounds. Such algorithms already exist. On the other hand, it recognizes the language family.

A supplemental feature is a type of feedback structure that allows the success and level of speech intelligibility to be checked in near real-time to then adjust the parameters of the masking sound to be generated.

In the implementation and adaptation of the vibroacoustic system according to the invention and method for voice masking the sound transmission of lightweight partitions there are several design options. Thus, in the case of mutual danger of interception between two adjacent rooms, a bidirectional system is possible which functions separately or in a coordinated manner in both directions. All mentioned

Features can be transferred to so-called flanking components, e.g. on walls that are perpendicular to the actual separation component 1 and allow longitudinal sound transmission between rooms. Analogously, the flanking applies

Sound transmission but also for lines, channels, pipes, etc., which are stimulated by the sound in the room 2 transmitted speech signals in the building. In this case, that is

Separating component 1 of a multi-shell duct casing with otherwise identical structure of the elements of the vibroacoustic system according to the invention and method for voice masking. Fugues are a particular challenge, in particular inevitable functional joints, z. B. in doors. However, both in a fundamentally advantageous overlapped embodiment of door and frame as well as in a special element for covering the joint, the inventive vibro-acoustic system and method for voice masking can be used again. In this case, one on the actual separation component 1, ie z. As the door, coordinated design provided.

The application of the vibroacoustic system and method according to the invention for speech masking is not limited to buildings, but can also be applied in

Vehicles and means of transport of different kinds are used advantageously.

Claims

claims

1 . Method for reducing the intelligibility of a speech signal with the following steps:

a) detection of amplitude, time and frequency of the speech signal (8);

b) generating a masking sound (9) which is so dependent on the amplitude, time and frequency of the speech signal (8) that by overlapping the speech signal (8) with the masking sound (9) an incomprehensible overlay sound (12) is formed, wherein the overlay sound (1 2) does not have a much higher volume than the speech signal, wherein the generation of the masking sound (9) is subject to a random change and wherein the speech signal (8) is divided into intervals, characterized in that the length of the intervals according to a

Random function can be changed randomly.

2. The method according to claim 1, characterized in that the time intervals in a range of 20 ms to 200 ms, in particular in a range of 50 ms to 1 00 ms, are changed randomly.

3. The method according to any one of the preceding claims, characterized in that an insulation and / or attenuation of the speech signal (8) in the generation of the masking sound (9) is taken into account, in particular a masking sound (9) is provided with low volume, if the Superposition sound (1 2) from the superposition of the masking sound (9) with the insulated and / or attenuated speech signal (1 1) takes place.

4. The method according to any one of the preceding claims, characterized in that the masking sound (9) is formed by temporal inversion of the short intervals.

5. separation component (1) for separating a room (2) from a room environment (3) with sensors (4), actuators (5) and a signal processing unit (6), for influencing the sound transmission, characterized in that the

Separating component (1) comprises the following components:

Sensors (4) for detecting the voice signal (8) coming from the room (2); Actuators (5) for generating a masking sound which is dependent on the amplitude, time and frequency of the speech signal (8);

a signal processing unit (6) which can process the signals detected by the sensors (4) so that the actuators (5) can generate such a masking sound (9) that by superposing the masking sound (9) with the speech signal (1 1) in the room environment an incomprehensible

Superposition sound (12) is formed without the superposition sound (12) must have a much higher volume than the voice signal (1 1), the sensors (4) and the actuators (5) are spaced, so that the

Speech signal (8) has a certain transit time, until it comes to the superposition with the masking sound (9), wherein the signal processing unit (6) is adapted to delay the masking sound (9) in time with respect to the speech signal (8) to run times of the speech signal ( 8) in the separating component (1).

6. Separating component according to claim 5, characterized in that the separating component

(I) is formed so that the generation of the masking sound (9) is subject to a random change.

7. Separating component according to claim 5 or 6, characterized in that the

Separating member (1) is adapted to perform a method according to any one of claims 1 to 4.

8. Separating component according to one of claims 5 to 7, characterized in that the separating component (1) is multi-layered or multi-shelled.

9. separating component according to one of claims 5 to 8, characterized in that the separating component is arranged so that the superposition of the masking sound

(II) takes place with the speech signal (8, 1 1) within the separating component (1), so that a body sound is formed in an area (10) of the separating component (1) facing the room environment (3), the incomprehensible overlay sound (12) into the room environment (3) radiates.

1 0. Separating component according to one of claims 5 to 9, characterized in that the separating component (1) is arranged so that in the room environment (3) a Superposition of the passing through the separating member (1) share of

Speech signal (1 1) with the masking sound (9).

1 1. Separating component according to one of claims 5 to 10, characterized in that the sensors (4) are designed as vibroacoustic sensors.

1 2. Separating component according to one of claims 5 to 1 1, characterized in that the sensors (4) on the space (2) facing side (7) of the separating component (1) or in this area (7) and / or the Actuators (5) at the

Room environment (3) facing side (10) of the separating component (1) or in this area (7) are mounted.

1 3. Separating component according to one of claims 5 to 12, characterized in that the separating component (1) is designed as a door between the space (2) and room environment (3).

14. Separating component according to one of claims 5 to 13, characterized in that the separating component (1) is designed as a solid component.

1 5. Separating component according to one of claims 5 to 14, characterized in that the separating component (1) is formed by a wall-like component with a facing shell.