System and method for processing microphone signals

Abstract

The invention relates to a system for processing microphone signals, the system comprising: a first processing device for providing a first group of individual microphone signals and a second processing device for providing a second group of individual microphone signals, wherein the first processing device is designed to transmit the first group of microphone signals, via a first radio link, from the first processing device to the second processing device, which is designed to receive the first group of microphone signals, and wherein the second processing device is designed to transmit the second group of microphone signals, via a second radio link separate from the first radio link, from the second processing device to the first processing device, which is designed to receive the second group of microphone signals.

Description

[0001] title

[0002] System and method for processing microphone signals

[0003] Description

[0004] Technical field

[0005] The invention relates to a system and method for processing microphone signals.

[0006] background

[0007] WO 2021 / 023771 A1 discloses a binaural system and a method for improving the speech of one or more desired speakers. In this system, the microphone signals recorded on the respective side of the head are transmitted via a first communication channel from the left to the right side of the hearing aid system and vice versa using a communication interface on both the right and left sides of the system. This type of microphone signal transmission has proven disadvantageous because, for example, unpredictable latencies in the mutual transmission of the microphone signals mean that the continuous function of the binaural hearing aid system cannot be permanently ensured. Furthermore, the known system uses a second communication channel to receive directional information regarding audio signal sources from a portable user device.

[0008] Against this background, the invention has set itself the task of creating an improved system and an improved method.

[0009] Summary of the invention

[0010] This object is achieved by a system according to claim 1.The subject matter of the invention is therefore a system for processing microphone signals, comprising: a first processing device for providing a first group of individual microphone signals and a second processing device for providing a second group of individual microphone signals, wherein the first processing device is designed to transmit the first group of microphone signals via a first radio connection from the first processing device to the second processing device, which is designed to receive the first group of microphone signals, and wherein the second processing device is designed to transmit the second group of microphone signals via a second radio connection separate from the first radio connection from the second processing device to the first processing device, which is designed to receive the second group of microphone signals.

[0011] This object is further achieved by a first method according to claim 24. The invention therefore relates to a first method for operating a system for processing microphone signals, wherein the first method comprises the following method steps, namely: providing a first group of individual microphone signals with a first processing device and providing a second group of individual microphone signals with a second processing device, wherein with the aid of the first processing device, the first group of microphone signals is transmitted via a first radio connection from the first processing device to the second processing device, which is designed to receive the first group of microphone signals,and wherein, with the aid of the second processing device, the second group of microphone signals is transmitted via a second radio connection separate from the first radio connection from the second processing device to the first processing device, which is designed to receive the second group of microphone signals.

[0012] This object is further achieved by a second method according to claim 25. The invention therefore relates to a second method for operating a processing device, the second method comprising the following method steps, namely: providing a group of individual microphone signals present at the processing device via a first radio connection and receiving a group of individual microphone signals present at another processing device via a second radio connection separate from the first radio connection. This object is further achieved by a computer program product according to claim 26. The invention therefore relates to a computer program product with software code that is adapted to execute the method steps according to the second method on programmable hardware of the processing device.

[0013] The measures according to the invention have the advantage that, unlike known measures, the transmission or exchange of microphone signals between the two processing devices takes place via an individual radio connection. This situation arises in which the two processing devices are not connected to each other via a wired connection, but are positioned at a distance from each other without a wired connection between them. Rather, they are connected to each other via a wireless connection.

[0014] To exchange the groups of microphone signals, a separate radio connection is used for each group of microphone signals, so that each radio connection only needs to transmit a single group of microphone signals unidirectionally. The respective radio connection is therefore prioritized for the respective group of microphone signals, and the full available bandwidth can be used for this unidirectional transmission of the payload data representing the microphone signals. Of course, this does not preclude the bidirectional exchange of additional control data required for the unidirectional transmission of the payload data between the two processing devices communicating via the respective radio connection.

[0015] The use of two separate radio connections also ensures the highest possible data availability per radio connection, because each radio connection is used for only one transmission direction, and the respective radio connection is therefore unaffected by the other transmission direction. This means that the full bit rate, i.e., the maximum connection speed for unidirectional transmission of payload data, is available for each radio connection.

[0016] The unidirectional transmission also has the advantage that the required amount of control data is kept to a minimum, meaning that the so-called "handshake" can be implemented as simply as possible.

[0017] It also dramatically increases the security of data transmission against unauthorized access, because the use of only one transmission direction per radio connection eliminates many otherwise possible entry points for hacker attacks right from the start.

[0018] In summary, the measures discussed ensure that the microphone signals required for further processing are exchanged between the two processing devices with the highest possible quality and within the shortest possible time without interference from each other, thus virtually eliminating the problem of unpredictable latencies mentioned in the introduction. Only this circumstance ensures that the entire sound field information, captured on the one hand by the first group of microphone signals at a first location and on the other hand by the second group of microphone signals at a second location, is available to the respective processing device for further processing as quickly as possible.This provides a solid basis for sufficient time for further (predominantly digital) processing of the two groups of microphone signals in both processing devices, so that ultimately the entire processing chain, especially including complex processing steps, can be executed so quickly (e.g., with a total processing time in the range of 1 to 10 milliseconds) that the processing time required for this is imperceptible to humans. Therefore, only the fact that the exchange of the groups of microphone signals occurs extremely quickly and reliably with the help of the discussed measures ensures that real-time behavior is present for the entire processing chain, i.e., the processing time incurred throughout the entire processing chain is so short that the processing time is no longer perceptible to humans.The rapid and reliable exchange of microphone signal groups thus even allows for the implementation of complex and computationally intensive further processing steps, simply because more time is now available for this. Further, particularly advantageous embodiments and developments of the invention will become apparent from the dependent claims and the following description.

[0019] The first group of individual microphone signals occurs directly at the first processing device and is generated with the help of at least two microphones positioned there, which convert the sound incident upon them into microphone signals, thereby generating at least two individual microphone signals. These at least two microphone signals are available via wired transmission at the first processing device. The second group of individual microphone signals is received at the first processing device via radio from the second processing device and is thus available via radio transmission at the first processing device.

[0020] The second group of individual microphone signals occurs directly at the second processing device and is generated with the help of at least two microphones positioned there, which convert the sound incident on them into microphone signals, thereby generating at least two individual microphone signals. These at least two microphone signals are available to the second processing device via a wired connection. The first group of individual microphone signals is received by the second processing device via radio from the first processing device and is thus available to the second processing device via radio.

[0021] The system can, for example, be used in a studio recording system, a concert recording system, or even a conference audio system. A preferred embodiment of the system is a binaural hearing aid system, wherein the first processing device is designed to be worn at the position of the left ear of a person's head, and the second processing device is designed to be worn at the position of the right ear of the person's head. In this embodiment, the processing devices can be referred to as so-called "hearables" or "intelligent" headphones or hearing aids. The hearing aid system therefore usually has two hearing aids, one for the left ear and one for the right ear of a person. The hearing aids are then designed orshaped so that they can be worn on the ear (also known as "behind-the-ear hearing aids", also known as "receiver-in-canal (RIO) hearing aids") or in the ear (also known as "in-the-ear hearing aids"). So-called in-the-ear hearing aids can be so slim that they can be inserted into the external auditory canal. Behind-the-ear hearing aids can be curved so that they can be positioned between the auricle and the skull, i.e. that they partially encompass the auricle or ear at the upper and / or back edge and are thus reliably held there.

[0022] A binaural hearing aid system is generally understood by experts to be a hearing aid system that enables a person with hearing loss to hear as naturally as possible in both ears. To achieve this, a binaural hearing aid system amplifies the volume unilaterally (i.e., for one ear) or bilaterally (i.e., for both ears) according to the impairment of each ear, thus achieving a natural sound perception for the wearer of the binaural hearing aid system. In addition, binaural hearing aid systems can be designed to provide functions such as amplifying certain sounds, localizing certain sounds, suppressing certain disturbing sounds, or even acoustically focusing on specific voices or sounds.

[0023] The binaural hearing aid system according to the invention presented here is designed to record acoustic audio signals on both sides of the head and to transmit them as microphone signals from both sides of the head to the other side of the head, to process them and, depending on the needs of the person using the binaural hearing aid system, to transmit a signal individually modeled for the corresponding ear to one ear or to both ears.

[0024] The signal thus transmitted to the respective ear is reproduced there in a manner perceptible to the person. For this purpose, it has proven advantageous for at least one processing device, preferably both processing devices, to each have an output module, in particular a loudspeaker and / or an output module for controlling an implant. The implant can be, for example, a cochlear implant or a brainstem implant. However, a loudspeaker is preferably used as the output module.

[0025] In general, the system's output module can comprise multiple sound output devices, such as loudspeakers. Two of these can be provided for binaural or stereo output, or more than two can be provided for spatial sound output at a higher order.

[0026] For the mutual transmission of the groups of microphone signals, each of the processing devices can have exactly one radio module, with a first radio channel of a frequency band being used for the first radio connection, and a second radio channel different from the first radio channel (in the same or a different frequency band) being used for the separate second radio connection. In the respective radio channel, the respective radio module therefore operates either as a transmitter or as a receiver. It must therefore switch back and forth, or "multiplex," between these two functions.

[0027] In a preferred embodiment, however, the first processing device in the system has a first radio module for transmitting the first group of microphone signals, and the second processing device has a second radio module for receiving the first group of microphone signals. Furthermore, according to this preferred embodiment, the second processing device has a third radio module for transmitting the second group of microphone signals, and the first processing device has a fourth radio module for receiving the second group of microphone signals. Thus, each processing device has a separate radio module for transmitting and a separate radio module for receiving, so that the two separate radio connections can be handled without switching back and forth between the receive mode and the transmit mode in the respective module.The first radio connection exists between the first radio module and the second radio module, whereas the second radio connection exists between the third radio module and the fourth radio module. Each radio connection can use different frequency bands. Preferably, a single frequency band is used, but systematically different channels are used for transmitting the respective group of microphone signals.

[0028] To provide the wireless connection, various wireless technologies and communication standards generally known to those skilled in the art, such as Near Field Communication (NFC), ZigBee, Wi-Fi, or even a proprietary protocol, can be used. The wireless modules are designed according to the respective wireless technology, and special software (driver software) that provides the respective protocol may be used.

[0029] It has proven particularly advantageous when the wireless connections are Bluetooth wireless connections. For this purpose, the wireless modules are designed as Bluetooth wireless modules.

[0030] Furthermore, it has proven particularly advantageous that for each processing device, one radio module is configured as a Bluetooth master radio module, with the aid of which the respective group of microphone signals is sent from the respective processing device to the other processing device, and one radio module is configured as a Bluetooth slave radio module, with the aid of which the respective group of microphone signals is received from the other processing device.

[0031] In a particularly preferred embodiment, each processing device has two Bluetooth 5.2 modules, in particular implemented as Bluetooth Low Energy Audio Modules (BT-LE Audio Modules for short). The Multi-Link protocol is used. This allows several Bluetooth radio modules to be used together in groups. The Bluetooth radio modules can each be the master of a group and thus serve as an audio broadcaster, or be configured as a slave and thus receive the audio signals from the master. Each processing device thus contains two radio modules. One is defined as the Bluetooth master, to which the two microphones are connected via a wired connection. The second radio module is defined as the Bluetooth slave and receives the microphone signals from the Bluetooth master of the other processing device via Bluetooth radio.

[0032] Bluetooth 5.2 has proven particularly advantageous because it supports the so-called "Auracast" functionality, which is very interesting because it allows devices, including the processing units, to transmit and receive public audio streams, thus enabling a method of interaction with an external app.

[0033] The master / slave configuration ensures that the framework for the activities to be performed jointly is established. The master dictates these rules or framework conditions. Tasks can thus be delegated to any number of slaves to achieve coordinated task distribution. A set of rules that applies to both sides is defined to enable collaboration. This is precisely what the master / slave configuration is suitable for. The master / slave configuration also ensures that a pair of antennas is available for unidirectional transmission, thus avoiding "multiplexing" and the like.

[0034] In principle, the respective radio connection can be re-established for each communication requirement. However, to ensure the best and fastest possible availability of the radio connection, it has proven particularly advantageous for the two processing units to be designed to provide a permanent first and second radio connection. This has the effect of ensuring that the system is always ready to receive or transmit, because the necessary connection only needs to be established initially and is then continuously available.

[0035] In the case of one module per processing device, this module must provide both the receive and transmit functions "multiplexed" as previously discussed.

[0036] In the case of two modules per processing device - as previously discussed - "multiplexing" is avoided because an individual module of the first processing device and an individual module of the second processing device provide this combination of transmit and receive readiness in a permanently paired configuration.

[0037] In this configuration, the system realizes a wireless, phase-synchronous microphone signal array for the two processing devices through its constant data exchange via the respective module pairs, whereby a first module pair is given by the first radio module and the second radio module and a second module pair is given by the third radio module and fourth radio module, between the two processing devices, so that at each processing device all two groups of microphone signals are available in phase-synchronization for further processing beyond the pure exchange of the microphone signals.

[0038] In addition to the measures mentioned above, it can also be helpful for the respective (digitized) microphone signals to also contain synchronization data, for example, or for this data to be included in the data stream of the microphone signals. This allows the temporal relationship between the individual microphone signals to be reconstructed in real time using this data. The necessary synchronization data or information can also be transmitted separately from the microphone signals via the respective radio connection. If necessary, another separate radio connection can also be used for this purpose.

[0039] Furthermore, it has proven advantageous not only to specify the temporal relationship between the microphone signals for their respective processing in the processing device, but also to synchronize their respective processing in the respective processing device across the processing devices in such a way that the output signals generated therefrom are transmitted via the output module in such a synchronous manner that deviations from this are indecipherable or perceptible to the human ear. Preferably, therefore, the two processing devices are synchronized with each other. For this purpose, the two processing devices can be operated with a processing synchronization signal that applies to both processing devices. They can therefore be operated with a common clock signal, or both processing devices can be operated according to a common time base, etc.The availability of the processing synchronization signal for both processing devices can be achieved through radio communication between the two processing devices. The radio communication for transmitting the processing synchronization signal can be carried out in the same way as the transmission of the synchronization data. The synchronization data can also include the processing synchronization signal.

[0040] With regard to the further processing of the groups of microphone signals, it has proven advantageous that the first processing device in the system is configured to process the first group of microphone signals together with the second group of microphone signals received via radio link in a phase-synchronized manner, and that the second processing device in the system is configured to process the second group of microphone signals together with the first group of microphone signals received via radio link in a phase-synchronized manner. This ensures temporally symmetrical processing of the two groups of microphone signals per processing device, as well as across all processing devices in the entire system.Based on the previously discussed advantageous type of mutual transmission, sufficient time is now available for this type of further processing, with the respective computing power of each processing device being available for the individual signal processing of all microphone signals. In the present context, phase synchronization means that only simultaneously occurring sound events, which are captured in the form of microphone signals, are processed together. In the signal processing of all microphone signals, the distance between the two processing devices, which is essentially defined by the distance between the two ears, is advantageously taken into account.

[0041] As mentioned, the microphone signals are generated by individual microphones. It is advantageous if the first radio module is configured to receive at least a first microphone signal from a first microphone and a second microphone signal from a second microphone and to transmit them as the first group of individual microphone signals, and if the third radio module is configured to receive at least a third microphone signal from a third microphone and a fourth microphone signal from a fourth microphone and to transmit them as the second group of individual microphone signals. In this basic configuration, a three-dimensional interpretation of the entirety of the microphone signals can be performed.For this purpose, it is advantageous if the microphones are designed and oriented in such a way that, for each processing device, a directional characteristic of one microphone is essentially oriented (obliquely) forwards and a directional characteristic of the other microphone is oriented (obliquely) backwards. Thus, for example, the directional characteristic of the first microphone and the third microphone can be oriented (obliquely) forwards and the directional characteristic of the second microphone and the fourth microphone can be oriented (obliquely) backwards. Here, the orientation in the horizontal plane was primarily discussed. However, the orientations of the individual microphones can also be pivoted out of this plane. Combinations of the aforementioned orientations are also possible.If a hearing aid system with this microphone arrangement is worn by a person, with one processing device on the left ear and one on the right ear, a microphone with a directional characteristic for sound detection to the front left, back left, front right, and back right is available relative to the person's head. This enables, at least to a first approximation, three-dimensional detection of sound sources in the area surrounding the head and further processing in this context.

[0042] It should also be mentioned in this context that more than two microphones per processing device can be used to generate groups of microphone signals comprising more than two individual microphone signals, in particular with further orientations different from those mentioned so far, which results in an improved resolution for the three-dimensional detection of the sound field.

[0043] The processing of the microphone signals is preferably carried out by each processing device for all microphone signals available to it.

[0044] It has proven advantageous that each of the processing devices has a processing module which is designed to process the two groups of microphone signals.

[0045] This has the advantage that the function of radio communication between the processing devices is separated from the "data processing," i.e., the further processing of the microphone signals in the respective processing device. Therefore, in addition to the radio modules that directly receive and forward the respective microphone signals, there is preferably a separate processing module for further microphone signal processing. Such a processing module can be implemented by a discrete logic circuit, a microprocessor, a microcontroller, or similar.

[0046] Various generally known procedures, methods, or algorithms can be used to process microphone signals. For example, the microphone signals can be further processed using Dolby Atmos technology or Aureal 3-Dimensional technology, or A3D technology for short, or the Auro-3D format.

[0047] It has proven particularly advantageous that each of the processing devices is designed to process the two groups of microphone signals according to the Ambisonics format.

[0048] The major advantage lies in the fact that this format is widespread and therefore widely accepted. This makes it easy to use because predefined signal processing procedures exist. Ambisonics is therefore a fully spherical audio format generally familiar to experts and is used primarily in the B-format.

[0049] The use of the Ambisonics format also allows for simple and familiar use of the captured microphone signals. For example, they can be recorded and saved as recording data. This recording data can then be further processed within the system or by other devices. For example, the recording data can be used by so-called "(e.g., video and / or audio) content creators" to use the recorded recording data in their own (proprietary) processing environment.

[0050] Advantageously, in the context of the invention, the Ambisonics format is used in the further system-internal processing chain to enable the use of one or more virtual microphones. This allows the system to focus the directional characteristics of each virtual microphone on individual sound sources (e.g., devices, vehicles, or people) in the sound detection range purely on a software basis, i.e., without any additional external aids. This has proven particularly advantageous when using the system as a binaural hearing aid system to support hearing-impaired individuals.

[0051] The aforementioned basic configuration, consisting of two microphones per processing device, already forms the basis for audio signal processing according to the generally known first-order Ambisonics format. If necessary, a transformation of the microphone signals from one coordinate system to another may be necessary in order to make the preferred arrangement and / or orientation of the microphones accessible for use with the Ambisonics format in accordance with the convention. If more microphones are present than in the basic configuration, and if corresponding orientations or directional characteristics are present, higher orders of the Ambisonics format can also be processed. It should be noted that even when using other formats, the use of multiple microphones allows a description of a higher order of spherical harmonics.

[0052] The fully automatic or user-interaction-based three-dimensional interpretation of the microphone signals makes it possible to identify a specific sound source and its direction relative to the current orientation of the processing device. The sounds from this sound source can then be amplified or suppressed for the person depending on the setting or the characteristics of the sounds.

[0053] The direction of the sound source can be determined fully automatically using the process of "beamforming" (known in the context of Ambisonics). Beamforming generally refers to the orientation of a directional characteristic (of a virtual microphone known in the context of Ambisonics), and possibly also the modeling of the directional characteristic (of this virtual microphone).

[0054] If the sound source or system is moved through the room, the change in the position of the sound source relative to the system, or its orientation—which in the case of a hearing aid results from the orientation of the head—must be taken into account in order to maintain focus on the previously focused sound source and to amplify or suppress its sound. The aforementioned "beamforming" with a virtual microphone can also be used for this process—possibly even automated. This allows the sound field to be continuously scanned and the position of the "moving" sound source to be tracked.

[0055] However, since this is very computationally intensive, it has proven particularly advantageous that at least one of the processing devices, preferably both processing devices, has a position sensor for providing a position sensor signal, preferably an electro-mechanical position sensor, particularly preferably an acceleration sensor and / or a gyroscope, and is designed to process the microphone signals taking into account the position sensor signal.

[0056] If the position sensor is only present on one side, its position sensor signal can be communicated wirelessly to the other processing device so that it is also available for processing there.

[0057] The position sensor tracks all movements of the processing device and generates a position sensor signal dependent on this. Each position sensor is preferably coupled to the processing module, making the position sensor signal available there. The processing module thus receives the position sensor signal and can take it into account when processing the microphone signals.

[0058] This means, for example, that the voice of a person speaking can continue to be focused on and amplified over other noises, even if the person speaking and / or the person wearing the system changes position or orientation. This makes it possible, for example, to have a conversation while walking, with the person wearing the system shifting their gaze back and forth between the person speaking and the sidewalk. The position sensor signal, which describes the head rotation, allows the respective processing device or module to take this movement into account when evaluating the microphone signals and continuously maintain focus on the person speaking, thus continuously amplifying their voice over other noises.The data obtained by the position sensor, particularly the gyroscope and / or accelerometer, is used to determine the position or change in position of a sound source relative to the person using the system. The person's movement is thus compensated. This information can then be used, for example, to "fix" the sound source and specifically amplify or suppress the noise from this sound source, even if the person moves relative to the sound source.

[0059] In summary, the gyroscope measures rotation around the three spatial axes, and this measurement is incorporated into a rotation matrix that rotates the recorded sound field while maintaining the reference direction from the listener. This ensures that the virtual microphone always remains pointed at a source, regardless of the user's head orientation.

[0060] The "fixation" of the sound source can be done autonomously or automatically. For this purpose, for example, pattern recognition, particularly voice recognition, can be used to assess whether a sound source is a human voice or speech. This enables uncomplicated use of the system and thus simplifies communication with another person.

[0061] Furthermore, to stabilize the desired listening direction, one can simply attenuate microphone signals from microphones not oriented in the desired direction. Microphone signals from microphones oriented in the desired direction can also be amplified. These measures can also be used in combination and dynamically adjusted for each microphone, so that the desired listening direction is maintained despite head movements.

[0062] However, if several voices can be heard in a room and the direction from which they arrive is then automatically identified, it is advantageous if the user of the system has the option of directing his focus to a person or a sound source by means of user interaction or of specifically keeping a selection of people or sound sources in the acoustic focus.

[0063] However, a LIDAR sensor (Light Detection and Ranging or Light Imaging, Detection and Ranging), a radar sensor or an infrared sensor, etc. can also be used as a position sensor.

[0064] According to a further aspect of the invention, it is advantageous that at least one processing device, preferably both processing devices, has an input module for detecting a user interaction and the processing device is designed to process the microphone signals taking into account the detected user interaction.

[0065] For this purpose, the input module can have buttons, switches or keys that record a haptic user interaction and make this event electronically accessible for further processing by the respective processing device.

[0066] The input module can have a camera or be designed as a camera. Such an input module can, for example, detect a hand gesture as a user interaction. For this purpose, the processing device can be designed to recognize objects, in particular the hand or its movement, or the position or combination of finger orientations, in the captured image data using pattern recognition.

[0067] It has proven advantageous that the input module is designed to detect user interactions within the auricle or in front of the auricle or along the front edge of the auricle, in particular towards the upper edge.

[0068] This allows the system user to trigger user interaction with movements that are as natural as possible. To capture user interaction in this area, it is advantageous for a capture element of the input module to be designed to be worn in the ear. The input module can thus have a camera that captures user interaction in front of the ear from within the ear. This allows the user to control the system with simple hand gestures.

[0069] Particularly preferably, the input module has at least one touch bar or is designed as a touch bar.

[0070] Unlike buttons, switches, or keys, a touch bar doesn't require a fixed actuator to be moved. Instead, a simple touch of its touch-sensitive surface is sufficient, because the touch bar is designed for capacitive touch detection.

[0071] This enables a wide range of user interactions in a very small space, with the user interaction being felt by the interacting person because they have to touch the Touch Bar. User interactions can therefore be various sequences of taps and swipes in different directions at different locations on the Touch Bar. Thus, only minimal space is required to enable the user to perform many different inputs.

[0072] Within the touch-sensitive surface, touch zones on the touch bar can be defined using hardware or software, or even adjusted or modified during operation. This creates virtual, individually definable buttons that are assigned functions of the processing device, such as switching between predefined profiles (for controlling microphone signal processing). This allows for targeted and rapid selection of profiles or even automated, situation-specific adjustment of the profile selection (for a signal processing chain for processing the microphone signals).

[0073] The touch bar is preferably also designed to detect user interactions within the auricle, in front of the auricle, or along the front edge of the auricle, in particular towards the upper edge, and is preferably positioned within the auricle for this purpose. The touch bar is preferably shaped to suit the respective anatomy and positioned at the front edge of the auricle, where uncomplicated and intuitive touching from the side is possible. The arm does not need to be raised as far, and the fingers do not need to be bent as much, than would be the case if the outer part of the touch bar were positioned along the back of the auricle, where it could only be touched from behind. This is particularly important for older people, whose mobility is usually limited.

[0074] In order to enable easy adaptability to the individual anatomical conditions of the ear, it has proven advantageous for a housing, on which a touch-sensitive surface of a first touch bar is accessible, to be connected at least on one side, but preferably on both sides, to a deformable, in particular plastically deformable, connecting element. This allows the position or orientation of the touch-sensitive surface to be specifically changed or adjusted. The housing is preferably tubular or rod-shaped so that the touch-sensitive surface is easy to position and easy to touch. If the touch-sensitive surface extends essentially along the entire length of the housing (apart from an enclosing edge), the largest possible area of ​​the touch-sensitive surface can be provided, so that it can subsequently also be used for complex functional controls orMenu functions that involve more than just tapping the touch-sensitive surface are available.

[0075] It is particularly advantageous that the deformable connecting element, as a first connecting element, connects a housing of a signal-emitting part of the output module to a housing on which the touch-sensitive surface of the first touch bar is accessible.

[0076] Preferably, the shape of the connecting element follows a continuous transition from the shape of one housing to the shape of the other housing. The shape is thus spatially designed and, in contrast to commonly used wire-like connections, can therefore absorb forces other than tensile forces, in particular absorbing bending loads and transferring them between the adjacent elements (output module / input module).

[0077] This allows for reliable positioning of the sound-emitting part of the output module at the position where the output should be most effective. For example, the loudspeaker for sound output can be placed there in a stable manner in the ear. At the same time, the element of the input module intended for interaction (touch-sensitive surface of the first touch bar) can be placed in the designated location, for example in or in front of the auricle. This also ensures free accessibility of the element of the input module intended for interaction, because it can be positioned in a variety of positions. The two housings thus support each other, while both the signal-emitting part of the output module and the touch-sensitive surface of the first touch bar can be positioned to suit the individual characteristics of the ear.

[0078] Furthermore, it has proven advantageous that the processing module has a housing, in particular a rigid housing, which housing is designed to be worn behind the auricle, which housing is particularly adapted to the shape of the auricle from the upper area to the rear area of ​​the auricle. This provides sufficient space to accommodate the

[0079] Processing electronics are provided. At the same time, this provides reliable protection against mechanical impact. The rigid housing also provides an optimal location for attaching or accommodating the at least two microphones because their position and orientation (in relation to one another) are thus unchangeably fixed. The rigid housing also provides optimal support for an additional touch-sensitive surface of a second touch bar of the input module, with the help of which, for example, the global volume of the system as well as individual volume for the two processing devices can be adjusted. Preferably, this additional element of the input module is assigned one or more functions that are required less frequently than those of the more easily accessible first touch-sensitive element of the input module.However, the bracket-like shape of the housing also means that the entire processing device can be reliably hung on the ear.

[0080] According to a further aspect, the housing of the processing module is connected on at least one side to a deformable, in particular plastically deformable, connecting element. This additional connecting element, with its deformability, also allows for easier positioning and alignment of the processing module in the holding position between the auricle and the skullcap.

[0081] Preferably, the deformable connecting element, as a second connecting element, connects the housing of the processing module to the housing on which the touch-sensitive surface of the first touch bar is accessible. This configuration makes it possible to establish a clamping effect that encompasses the auricle essentially on both sides, resulting from the interaction of the processing module with those parts of the processing device that face the face, i.e., essentially the housing on which the touch-sensitive surface of the first touch bar is accessible.

[0082] In particular, the use of the two flexible connecting elements with the housing positioned between them, on which the touch-sensitive surface of the first touch bar is accessible, comes into play here in order to create a structure that wraps around the auricle starting from the ear canal, the clamping effect of which is optimized by the two flexible connecting elements and enables operability from the front. Particularly preferably, the touch-sensitive surface of the first touch bar is held between the two deformable connecting elements. This is particularly because the two deformable connecting elements, once shaped, essentially retain this shape, i.e. are plastically deformable. However, the two connecting elements can also be elastically deformable. For example, they can be designed to have a basic shape that is tailored to the respective auricle.When putting on the hearing aid, the arch defined by the shape of the processing device can be expanded, making it easier to insert it between the auricle and the skull. One connecting element can also be plastically deformable and the other elastically deformable to provide both effects.

[0083] According to a further aspect, at least one of the processing devices has a wired connection, preferably a multi-pole connection, in particular a magnetically held connection, for connecting a further device or an additional module. This connection is preferably formed on a housing of the processing device. This connection is particularly preferably available at its free end, which, in the case of a processing device worn on the ear, is located behind the auricle or points downwards. This allows a flexible and variable expansion of the functionality of the respective processing device by the module or component connected to the connection.This allows the connection of at least one additional external microphone, a battery or charger, an external storage medium for recording microphone signals or for loading software or audio data, a wireless communication module for establishing a connection to another wireless network or enabling a connection to the Internet, for example to make a cloud solution available. A wireless transmission device can also be used as an additional module; this is logically paired with the system via the (magnetically held) connector. This transmission device can then be powered by another audio source or a control unit and can transmit audio data and / or configuration data to the system via its wireless connection.

[0084] According to a further aspect, one of the processing devices has a haptic module (also referred to as a haptic module) for emitting a haptically perceptible signal. This enables haptic signaling or haptic feedback in the vicinity of the ear, so that the haptically perceptible signal can be clearly assigned to a function or event related to the processing device. This can prevent potentially disruptive acoustic signaling. An example in this context is the successful establishment of an acoustic focus on a speaking person. A haptic signal can be provided, for example, as an electromechanically generated tapping or vibration, a pen movement, or the like.

[0085] According to a further aspect, one of the processing devices can be designed to receive and process an additional software application or audio output data representing an audio signal that can be emitted using the processing device. This makes it possible for the first time to load not only pure processing parameters or device settings, which are determined and applied by the audiologist when tuning the device to the respective application or wearer, into the processing device, but also executable program code, e.g. in the form of an application (this can be system plugins or user-defined plugins that influence or supplement the existing signal processing chain) and / or additional playable audio files. The additional application provides additional functionality or replaces an existing functionality with another, possiblyimproved functionality.

[0086] Examples of playable audio files include calming or relaxing background noises such as crackling burning wood, the sound of a fireplace, or chirping birds or the rustling of leaves in the forest, etc.

[0087] An example of additional functionality could be an alternative speech recognition algorithm—provided, for example, by a third-party provider (i.e., not directly by the system manufacturer)—where the alternative algorithm can be used either in addition to or instead of an existing algorithm. It can also be understood as a replacement functionality, for example, where an existing algorithm is replaced by a new one.

[0088] The additional software application or the audio output data can be loaded into both processing devices. Particularly preferably, the two processing devices are configured to transmit the software application received only by one of the processing devices or the audio output data received only by one of the processing devices to the other processing device using a third radio connection. This provides autonomous or automatic content synchronization of the software installed on the two processing devices or the available audio data. Preferably, content synchronization is performed first, before the microphone signals are processed on both processing devices using the software application or the audio output data. This ensures consistent functionality within the system.The third radio connection may preferably be a separate radio connection or be provided by the first radio connection or by the second radio connection.

[0089] It has also proven particularly advantageous if at least one of the processing devices is designed to communicate with a mobile user device via a fourth radio connection and to provide this user device with position information using the fourth radio connection, wherein the position information indicates the position of at least one sound source in relation to the position of the system, and to receive selection information from this user device using the fourth radio connection, wherein the selection information indicates which direction or directional range is selected for further processing of the microphone signals with respect to the position and / or orientation of the system. This allows the user of the system to specifically carry out one or more acoustic focusing operations using a visual interface. This functionality allows the use of a mobile user device, e.g.in the form of a mobile phone or a tablet computer to enable the user of the system to intervene in the signal processing chain on a visual basis.

[0090] In summary, the preferred application of the system is in the field of binaural hearing aids, particularly binaural Ambisonics hearing aids. However, the system can also be used for hearing therapy, earbuds, hearables (aural wearables), headphones, and assistive listening devices.

[0091] With regard to user interaction, an innovative user interface, in particular for a hearing aid, is further disclosed in its general form, i.e. regardless of the manner in which the microphone signals are transmitted between the processing devices or are processed there.

[0092] In this context, the processing device is designed to define at least two different menu areas in a two-dimensional interaction area of ​​the input module. The two-dimensional interaction area can be provided by the touch-sensitive surface discussed above.

[0093] The input module is further designed to detect a user interaction in the interaction area - in particular in a capacitive manner - and to provide this with the aid of interaction data which represent the interaction as such and / or a type of interaction.

[0094] The type of interaction can be a simple, brief touch (tap) on the interaction area, a relatively longer dwell on the interaction area, or a movement within the interaction area that touches the interaction area. For these types of interaction, a finger of a hand, a stylus, or something similar can be used.

[0095] By detecting a brief touch, the processing device can, for example, activate or execute the relevant menu item or the function associated with that menu item. By remaining in the interaction area for a relatively longer period, the processing device can, for example, switch from one menu level to the next.

[0096] By means of the movement taking place in the interaction area and touching the interaction area, the processing device can change a processing parameter corresponding to the width of the movement - e.g. to the respective menu item that was previously selected by simply tapping or activated by staying there for a longer time.

[0097] However, the processing device can also be configured to interpret the movement touching the interaction area in such a way that a menu area located at the beginning of the movement in the interaction area should be shifted or moved toward the end of the movement. This can be used for interactive regrouping of menu areas within the interaction area in order to adapt the position of the menu area(s) to the user's requirements.

[0098] However, the processing device can also be configured to interpret the movement affecting the interaction area in such a way that a first menu area located at the beginning of the movement in the interaction area is to be shifted or moved to another second menu area located at the end of the movement. According to a preferred embodiment, the two menu areas can be assigned different profiles—specifically, a first profile and a second profile—for the signal processing chain used in the respective processing device to process the two groups of microphone signals. Depending on the detected movement, the values ​​or settings of the signal processing parameters of the first profile are changed to those values ​​or settings of the signal processing parameters of the second profile.

[0099] This change in the values ​​or settings can occur continuously or quasi-continuously in the case of continuously changing values ​​or settings, whereby the degree of change can scale with the extent of the detected movement. In the case of values ​​or settings that can only be changed gradually or abruptly, the change to be made can occur when a threshold value of the detected movement is reached. In principle, such a profile can contain a mixture of both types of values ​​or settings.

[0100] When the change discussed occurs, the processing device converts the signal processing according to the first profile to signal processing according to the now existing changed values ​​or settings. The values ​​and settings can assume intermediate states between the limits of the values ​​or settings defined by the two profiles. Only if the detected movement utilizes the entire available length / width / diagonal, etc., of the areal interaction area or corresponds to a predefined movement range or shape, is a one-hundred percent switch made from the first profile to the second profile. Otherwise, the aforementioned intermediate states are used.

[0101] In the event that there is no corresponding parameter in the second profile for a signal processing parameter in the first profile, the signal processing parameter in question remains unchanged until a complete switch from the first profile to the second profile has occurred, or a percentage threshold (e.g., 25%, 50%, or 75%) of the maximum movement has been reached, or the essentially continuously changing values ​​or settings have been changed according to a threshold (e.g., 25%, 50%, or 75%). This prevents the effect of the parameter available only in the first profile from taking effect too abruptly. The same procedure can, of course, also be followed with a signal processing parameter in the second profile for which there is no corresponding parameter in the first profile.

[0102] The processing device can also implement the change in the signal processing parameters to be made according to the movement in real time in the signal processing chain, so that the result of this change in the signal processing parameters is immediately perceptible.

[0103] With the help of these measures, the user can intervene deeply in the function of the signal processing chain even during operation. In addition, a hearing test can be carried out and evaluated, and deficits can be identified because the input module allows for a wide variety of interactions. The system equipped with the technical measures described can also enable the sharing of an audio data stream generated from the microphone signals with another external device in real time. This can be done, for example, using a wireless connection to the external device or a wired connection via the (magnetically held) connector. This functionality allows another person to use the other device to experience the underlying audio experience of the person wearing the system live. This functionality can also be used to analyze the audio experience, i.e. the audio data stream, on the other device in real time.

[0104] The system, in particular the processing devices, can also each have a data storage device with which the generated microphone signals are stored in raw data format or pre-processed, continuously or limited to a temporal report of the occurrence of an audio event. Correspondingly, the settings or parameters used in the signal processing chain to create the audio data stream from the microphone signals are also stored. If the system user experiences problems with audio perception, targeted, systematic adjustment or modification of the settings or parameters can be used to determine which values ​​for the settings or parameters improve the user's audio perception.The stored data allows, in particular, incremental and repetitive adjustment of settings and parameters in order to determine the appropriate parameters for one and the same audio experience. This is particularly important because personal audio perception is often subjective, making it difficult for the person to communicate this objectively to an audiologist, for example. Furthermore, the audio data streams corresponding to the respective audio experience, which were created using a wide variety of settings and parameters or their modifications, can also be saved. This allows the user to narrow down the settings and parameters that are suitable for them through comparative listening (i.e., sequential playback of the audio data streams). This in turn enables the audiologist or a software-based audiological application to find the optimized set of settings or parameters.

[0105] Finally, it should be generally mentioned that the electronic devices discussed (both within the system and outside the system, such as smartphones, tablet computers, etc.) naturally contain electronics. The electronics can be discrete or implemented using integrated electronics, or even a combination of both. Microcomputers, microcontrollers, and Application Specific Integrated Circuits (ASICs), possibly in combination with analog or digital electronic peripheral components, can also be used. Many of the device functionalities mentioned are implemented – possibly in conjunction with hardware components – using software that runs on an electronics processor. Devices designed for radio communication usually have an antenna configuration for transmitting and receiving radio signals as part of a transceiver module.The electronic devices can also have an internal electrical power supply, which can be implemented, for example, with a replaceable or rechargeable battery. The devices can also be powered wired, either via an external power supply or via "Power over LAN."

[0106] These and other aspects of the invention are apparent from the figures discussed below.

[0107] Short character description

[0108] The invention is explained in more detail below with reference to the accompanying figures using exemplary embodiments, to which, however, the invention is not limited. In the various figures, identical components are provided with identical reference numerals. They show schematically:

[0109] Fig. 1 a head of a person with a binaural hearing aid system with one processing device at each ear of the head, Fig. 2 direction indication for the binaural hearing aid system in the horizontal plane related to the head,

[0110] Fig. 3 Directional characteristics of the microphones used in the binaural hearing aid system, Fig. 4 a block diagram of a binaural hearing aid system, Fig. 5 a signal processing chain used in the processing devices,

[0111] Fig. 6 structural details of the processing equipment in an exploded view,

[0112] Fig. 7 the processing device with electronic components or modules accommodated in its housing and additional expansion modules that can be connected to it,

[0113] Fig. 8 shows a dynamic interface of the processing devices realized with the help of two touch bars,

[0114] Fig. 9 an exemplary sound source arrangement in the environment of the binaural hearing aid system,

[0115] Fig. 10A - 10B show the effect of using the binaural hearing aid system in the exemplary sound source arrangement according to Fig. 9, Fig. 11A - HD show the use of an optional app on a mobile user device when using the binaural hearing aid system in a further sound source arrangement and

[0116] Fig. 12 a server-supported or cloud-based functional extension of the binaural hearing aid system.

[0117] Description of the embodiments

[0118] Figure 1 shows a head 101 of a person 100 wearing a binaural hearing aid system 1, hereinafter referred to as system 1 for short, which has a first processing device 2 worn on the left ear 102 and a second processing device 3 worn on the right ear 103 in order to enable the person 100 to have an improved, particularly natural hearing experience with both ears 102, 103. The designations front V, back H, left L, and right R correspond to the natural direction with respect to the head 101 of the (standing or sitting) person 100, so that the face is oriented towards the front V, the left ear 102 towards the left L, and the right ear 103 towards the right R. The same applies to the usual direction indications above O and below U. A radio connection is also indicated between the processing devices 2 and 3, which will be discussed in more detail below.

[0119] Figure 2 shows the four directions V, H, R, and L relative to the head 101 of the person 100, viewed from above, with the face of the person 100 facing forward V, corresponding to the degree of 90°. The degree is indicated in the horizontal plane, which is positioned essentially at the level of the ears 102, 103. The head 101 is indicated circumferentially and marked with a cross 200, with the longest section of the cross 201 oriented from the center of the head 101 toward the nose. This is mentioned here because this type of representation is also used in the subsequent figures.

[0120] Each of the processing devices 2 and 3 shown in Figure 1 has two microphones 10 and 11 or 12 and 13 (not shown, but see, for example, Figures 3 and 4), so that sound detection is possible in all four directions V, H, L and R, as shown in Figure 2. The main detection directions of the four microphones 10-13 shown in Figure 2 are indicated here by the directions "left-front" LV, "right-front" RV, "left-rear" LH and "right-rear" RH.

[0121] The directional characteristics LVC, LHC, RVC, and RHC of the microphones 10-13 shown in Figure 3 are (three-dimensionally) cardioid-shaped and extend essentially symmetrically around these main detection directions LV, RV, LH, RH. The directional characteristics LVC, LHC, RVC, and RHC are defined by the type of microphone 10-13 as well as the positioning and orientation of the microphone 10-13. The directional characteristics LVC, LHC, RVC, and RHC essentially cover the entire area around the head 101 of the person 100 and thus allow all-round detection of the sound arriving at the head 101. It should only be mentioned at this point that the microphones 10-13 are shown detached from the head 101 in order to make them visible. In fact, they are integrated into the processing devices 2 and 3, as can be seen in Figures 4, 6, 7, 8 and 12.

[0122] Figure 4 further illustrates the two processing devices 2 and 3 using a block diagram. The two processing devices 2 and 3 are battery-operated and therefore each have a rechargeable battery 25 that provides a supply voltage VCC relative to a reference potential GND.

[0123] The first processing device 2 has a first processing stage 4. The processing stage 4 is connected to a first Bluetooth radio module 6, briefly first radio module 6, and a fourth Bluetooth radio module 9, briefly fourth radio module 9, via a UART connection 17 for setting or programming the radio modules 6 and 9.

[0124] 9 and each with an I 2 S-connection 18 for separate transmission of audio data.

[0125] The abbreviation UART stands for "Universal Asynchronous Receiver Transmitter". The abbreviation I 2S is understood by experts as a serial interface for audio transmission developed by Phillips, which is also known under the term "Inter-IC Sound".

[0126] The first radio module 6 is connected via a wired connection (in this embodiment via its analog microphone signal inputs) to a first microphone 10 and a second microphone 11. The first microphone 10 and the second microphone 11 each generate an individual microphone signal MSI and MS2, respectively. The two microphone signals MSI and MS2 together form a first group G1 of microphone signals. The microphone signals MSI and MS2 arriving at the first radio module 6 are digitized for further processing using the first radio module 6.

[0127] The second processing device 3 has a second processing stage 5. The second processing stage 5 is connected to a third Bluetooth radio module 8, in short third radio module 8, and a second Bluetooth radio module 7, in short second radio module 7, via a UART connection 17 for setting or programming the radio modules 6 and 9, respectively, and via an I 2 S-connection 18 for separate transmission of audio data.

[0128] The third radio module 8 is connected via a wired connection (in this embodiment via its analog microphone signal inputs) to a third microphone 12 and a fourth microphone 13. The third microphone 12 and the fourth microphone 13 each generate an individual microphone signal MS3 and MS4, respectively. The two microphone signals MS3 and MS4 together form a second group G2 of microphone signals. The microphone signals MS3 and MS4 arriving at the third radio module 8 are digitized by the third radio module 8 for further processing.

[0129] At this point it should be mentioned that the respective group Gl or G2 of the microphone signals can also be fed to the radio modules 6 or 7 as a digitized audio data stream if appropriately designed microphones are used that support such a creation of the audio data stream.

[0130] Each of the radio modules 6 to 9 has an antenna configuration 14 and the usual transceiver electronics (not shown in detail) for Bluetooth radio communication.

[0131] During operation, a first radio connection 15 is established between the first radio module 6 and the second radio module 7, and the first group Gl of microphone signals is transmitted from the first radio module 6 to the second radio module 7. For this radio signal transmission, the first radio module 6 is used exclusively as a transmitter, and the second radio module 7 is used exclusively as a receiver.

[0132] During operation, a second radio connection 16 is established between the third radio module 8 and the fourth radio module 9, and the second group G2 of microphone signals is transmitted from the third radio module 8 to the fourth radio module 9. For this radio signal transmission, the third radio module 8 is used exclusively as a transmitter and the fourth radio module 9 is used exclusively as a receiver.

[0133] Furthermore, in this operating configuration, audio data representing the first group G1 of the microphone signals are transmitted by the first radio module 6 and audio data representing the second group G2 of the microphone signals are transmitted by the fourth radio module 9, each via the separate I 2S connections 18 to the first processing stage 4. Furthermore, in this operating configuration, audio data representing the first group G1 of the microphone signals are transmitted from the second radio module 7 and audio data representing the second group G2 of the microphone signals are transmitted from the third radio module 8, respectively, via the separate I 2S connections 18 are transmitted to the second processing stage 5. Thus, both groups G1 and G2 of the microphone signals, i.e., all microphone signals MSI to MS4, are available for further signal processing in both the first processing device 2 and the second processing device 3. The individual, unidirectional transmission via two separate radio connections 15 and 16—as discussed in the general part of the description—forms the basis for the further phase-synchronous signal processing of all microphone signals MSI to MS4 in each of the processing devices 2 and 3 in real time. In this exemplary embodiment, the four radio modules 6 to 9 are each implemented by a component from "MICROCHIP" with the designation "BM83 Bluetooth® Stereo Audio Module." The two processing stages 4 and 5 are implemented in this exemplary embodiment by a component from "NXP Semiconductors" with the designation "i.MX RT1160."For the sake of clarity, we have omitted any further (analog) components that may be required in conjunction with the aforementioned digital components. Experts can refer to the relevant data sheets for details.

[0134] Each of the processing devices 2 and 3 has a gyroscope 21a or 21b, respectively, with which position data LD is generated, which represents the position or the change in position of the respective processing devices 2 or 3. Each of the processing stages 4 and 5 is connected to the respective gyroscope 21a or 21b and takes the respective position data LD into account in the further processing of the two groups G1 and G2 of the microphone signals, so that position changes can be compensated for in this further processing.

[0135] Each of the processing devices 2 and 3 further comprises a touch-sensitive input module 20, which is divided, on the one hand, into a first input module controller 19a (implemented using a microcontroller) and a first so-called "touch strip" 20a connected to it, and, on the other hand, into a second input module controller 19b (also implemented using a microcontroller) and a second touch strip 20b connected to it. The two touch strips 20a and 20b form external manifestations of two touch bars, each of which has a touch-sensitive surface. The input module 20 can define freely assignable zones on the touch strips 20a and 20b, respectively, and thus divide the available area into menu items, for example, into an upper and lower half, so that the functions assigned to the menu items can be triggered by finger touch.The assignment of menu items to the zones can be predefined, changed or adapted depending on the operating state, or configured via an external app. The respective processing levels 4 and 5 are connected to the respective input module 20. The input module 20 is designed to capacitively detect the position of a finger of the person 100 on the respective touch strip 20a or 20b and to interpret it by means of the respective touch module controller 19a or 19b and to generate interaction data ID and provide it for the respective processing levels 4 or 5, where these interaction data ID are processed according to the currently valid menu assignment of the respective touch strip 20a and 20b. With the help of this menu assignment, for example,on the touch strip 20a and 20b define areas that are used to directly activate profiles, or define areas that are used to control the volume, or define areas that are used to align a virtual microphone, etc. Furthermore, it should be mentioned at this point that the interaction data ID can also be generated by means of an external interaction source, such as a smartphone, a tablet computer or a cloud-based software solution, and transmitted via a radio connection to one of the processing devices 2 or 3 or to both processing devices 2 and 3 in order to use them there for further processing.

[0136] The interaction data ID as well as the position data LD can be transmitted together with the microphone signals MSI and MS2 or MS3 and MS4 via the corresponding radio connections 15 and 16 to the other processing device 2 or 3, respectively, so that both processing devices 2 and 3 can be operated synchronously with each other or are simply mutually informed about the respective interaction or position.

[0137] The two processing devices 2 and 3 each have a visual signaling stage 26, which is essentially formed by an LED each, which is controlled by the respective radio module 6 or 8 to which it is connected. The visual signaling stage 26 serves, for example, to indicate transmission activity or the charge level of the battery 25 of the respective processing device 2 or 3.

[0138] Furthermore, each of the processing devices 2 and 3 has an output module 22, which is divided into an amplifier 22a or 22b and a loudspeaker 23a or 23b coupled thereto. The amplifiers 22a, 22b are an I2C Class D amplifier, which is connected via an I 2 C- connection is connected to the respective processing stage 4 or 5 and is designed to be supplied with digital audio data from there and to generate therefrom a correspondingly amplified output audio signal with which the loudspeaker 22a or 22b is controlled.

[0139] Furthermore, each of the processing devices 2 and 3 has a removable storage medium read / write stage 24, with the aid of which data can be written to a removable storage medium or read from the removable storage medium. This can be setting data, user data, audio data, or even application data, etc., which are made accessible to the system or the respective processing device 2 or 3 or retrieved from there. This can also provide memory expansion for computationally or data-intensive processing activities. This also allows executable applications to be physically provided, possibly even encrypted, in the system 1.

[0140] Reference numerals 27 to 29 are not used.

[0141] The signal processing chain 30 used in each of the processing devices 2 and 3 with the aid of the processing stages 4 and 5 is described below with the aid of Figure 5.

[0142] In this signal processing chain 30, the two groups G1 and G2 of the microphone signals MSI-MS4 are fed into a first signal processing stage 31 on the input side. All functions for Ambisonics-related signal processing are combined in the first signal processing stage 31, including or taking into account the position data LD and / or the interaction data ID that relate to Ambisonics-related signal processing. For example, an audio data stream structured according to the Ambisonics format is generated, the position data LD of the gyroscope 21a or 21b is taken into account, one or more virtual microphones are defined or controlled, etc. Using this functionality, the first signal processing stage 31 generates a first audio data stream ADI from the four microphone signals MS1-MS4, which is fed into a second signal processing stage 32.

[0143] The second signal processing stage 32 combines all functions for hearing curve adjustment-related signal processing, including or taking into account the interaction data ID relating to hearing curve adjustment. Hearing curve adjustment corrects the hearing curve of an impaired ear so that it approximates that of a healthy ear. A static equalizer is essentially used for this purpose. Using this functionality, the second signal processing stage 32 generates a second audio data stream AD2 from the first audio data stream ADI, which is fed into a third signal processing stage 33.

[0144] The third signal processing stage 32 combines all functions for hearing profile-related signal processing, including or taking into account the interaction data ID relating to the hearing profile settings. A hearing profile encompasses those audio signal processing parameters tailored to the respective ear or person, which improve or positively influence the intelligibility of speech, participation in a group discussion, listening to music, etc. If the respective hearing profile is activated, i.e., used in the signal processing chain, the intelligibility of individual speech is improved, participation in a group discussion is facilitated, the natural perception of music is promoted, etc.With this functionality, the third signal processing stage 32 generates a third audio data stream AD3 from the second audio data stream AD2, which is fed into a fourth signal processing stage 34.

[0145] The fourth signal processing stage 34 combines all functions for hearing aid function-related signal processing, including or taking into account the interaction data ID that affects the hearing aid function settings. These can include, for example, "quality of life improvements," such as additional noise filtering, echo / reverberation filtering, etc. Plugins that can be defined / downloaded by the user can also be integrated here. Using this functionality, the fourth signal processing stage 32 generates a fourth audio data stream AD3 from the third audio data stream AD3, which is delivered to the ear 102, 103 of the person 35 wearing the system 1 via the aforementioned output module 22.

[0146] Finally, for the sake of clarity, it should be mentioned that the signal processing stages 31 to 34, which are shown as structural blocks, can essentially be based on software modules, although it is of course also possible to use hardware that is optioned for the respective function, possibly a programmable one in combination with software.

[0147] The following discusses the physical structure of processing devices 2 and 3 with the aid of Figure 6. The two processing devices 2 and 3 are structured as follows.

[0148] They have a loudspeaker capsule 36 which is designed for use in the external auditory canal and accommodates the loudspeaker 23a or 23b in a housing.

[0149] They further comprise a first flexibly deformable connecting element 37, which is essentially formed from a body consisting of deformable plastic, in particular polyurethane plastic or silicone.

[0150] They further comprise a rigid first housing 38 which shows a first touch-sensitive part 39 of the first touch strip 20a, so that the touch-sensitive part 39 is accessible or touchable as unhindered as possible through the auricle on the first housing part when the processing device 2 or 3 is worn on the head, oriented laterally or forwards or oriented diagonally forwards.

[0151] They further comprise a second flexibly deformable connecting element 40, which is essentially formed from a body consisting of deformable plastic, in particular polyurethane plastic or silicone.

[0152] They further comprise a rigid second housing 41 which, on the one hand, is shaped to fit the outer auricle from its front to its rear region, in order to be received between the auricle and the skullcap, and which, on the other hand, is designed to be large enough to accommodate the remaining electronic components of the processing device 2 or 3, apart from the loudspeaker 23a or 23b and the first touch-sensitive part 39 of the touch strips 20a. These electronic components are connected by cable to the first touch-sensitive part 39 of the touch strips 20a and the loudspeaker 23a or 23b, which is not shown for reasons of clarity. The second housing part 41 therefore contains a processing module of the corresponding processing device 2 or 3. In this embodiment, the processing module comprises the respective processing stage 4 or 5, the two radio modules 6 and 9, respectively.8 and 7, two microphones 10 and 11, and 12 and 13, respectively, the respective gyroscope 21a and 21b, as well as the respective visual signaling stage (not shown here). Externally visible on the second housing 41 are a second touch-sensitive part 42 of the second touch strip 20b, wherein the second touch-sensitive part 42 is primarily used for volume adjustment, and a magnetically retained connector 43 or its contact field, which features six contact elements.

[0153] In the first processing device 2, the first microphone 10 or its sound inlet opening(s) is present on the second housing 41 at its front end, although this is not visible in the selected perspective of Figure 6. Furthermore, the second microphone 11 or its sound inlet opening(s) is visible on the second housing 41 of the first processing device 2.

[0154] Similarly, in the second processing device 3, the third microphone 12 or its sound inlet opening(s) is present on the second housing 41 at its front end, although this is not visible in the selected perspective of Figure 6. Furthermore, the fourth microphone 13 or its sound inlet opening(s) is visible on the second housing 41 of the first processing device 2.

[0155] Figure 7 is discussed below, in which the processing device 2 or 3 is shown assembled, in contrast to Figure 6. The areas of the rigid second housing part 41 are shown partially transparent or cut out in order to provide a view of the assemblies or electronic components arranged therein. In this illustration, the aforementioned integrated circuits as well as analog electronic components for the radio modules 6 and 9 or 8 and 7, for the input module 20 and for the processing stages 4 or 5 are arranged in an upper area 44A. In a rear area 44B, an electrical supply, which in the present case is implemented with two rechargeable button cell-like batteries 25, and a haptic module 45 (not yet shown) that can be controlled with the aid of the respective processing stage 4 or 5 are arranged.

[0156] Figure 7 also shows an optional expansion module 51 connected to the magnetically retained connector 43. This may include, but is not limited to, the following types of modules: a battery charging module 46, a microphone cube (not shown), a jack plug adapter 47, a USB adapter 48, a radio device 49 that can be logically linked to the system 1 and used to wirelessly transmit audio or settings data from another device to which the radio device 49 is then connected, e.g., via a USB port, to the system 1 or vice versa, a microphone extension 50, etc.

[0157] The rechargeable batteries 25 can thus be charged using the battery charging module. A docking station can also be provided, to which the two processing devices 2 and 3 can be coupled using the connector 43, so that the batteries 25 can be charged in the docking station.

[0158] In principle, System 1 can be designed to process Ambisonics according to a higher order than that provided by the microphones 1 to 13 permanently installed in System 1 (in this case, this is the first order). This circumstance allows additional microphones to be connected to port 43 and thus utilize the higher order without requiring any changes to the basic design of System 1.

[0159] With reference to Figure 8, the user interface integrated in system 1 is discussed below with an exemplary menu layout of the first touch-sensitive part 39 of the touch bar 20 for both of the processing devices 2 and 3 and the functions of system 1 that can be controlled thereby. In the present case, the menu layout is defined such that, in the second processing device 3, the upper half of the first touch-sensitive part 39 defines a first profile zone 53 for selecting a first profile for the signal processing chain 30, and the lower half of the first touch-sensitive part 39 defines a third profile zone 55 for selecting a third profile for the signal processing chain 30.Furthermore, the menu layout is defined such that in the first processing device 2, the upper half of the first touch-sensitive part 39 defines a second profile zone 54 for selecting a second profile for the signal processing chain 30 and the lower half of the first touch-sensitive part 39 also defines the third profile zone 55 for selecting the third profile for the signal processing chain 30.

[0160] The three profiles for the signal processing chain 30 summarize the respective signal processing settings or signal processing parameters that are to be used in the signal processing chain 30 depending on the activated menu.

[0161] Touching one of the profile zones (upper or lower half) of the respective touch-sensitive part 39 selects the respective associated profile 53-54 for application in the signal processing of the microphone signals MSI-MS4. This occurs in real time, while the person 100 is wearing the system 1 and the signal processing chain 30 is running through it. The profile transition between the currently active profile and the profile selected by touching the respective zone is preferably smooth, which is also referred to as "fading" and planned to take a certain amount of time, for example, one to three seconds, preferably approximately two seconds, so that a smooth transition to the selected profile, which is therefore to be used, is created. Abrupt signal processing changes, which could sometimes be interpreted as disruptive or malfunctioning, are thus reliably avoided.

[0162] However, the software of system 1, i.e. of each processing device 2 or 3, also allows a "hold and swipe" functionality, which is referred to in technical jargon as "drag and drop" functionality, which is advantageously used in the context of system 1 to mix profiles. The person first touches one of the profile zones 53 - 55, leaves the finger on it and then drags the finger into the adjacent profile zone before lifting the finger from the touch-sensitive element. The software interprets this in such a way that the profiles assigned to the two profile zones swiped over by the finger are to be mixed. If, for example, in the right-hand, second processing device 3, the finger starts with the first profile and swipes or drops downwards to the third profile.is deleted, the signal processing settings applicable to the signal processing chain 30 according to the first profile are mixed with those of the third profile. If the same is done with the left, first processing device 2, the signal processing settings of the second profile are mixed with those of the third profile. Furthermore, the degree of mixing (or in other words, the mixing ratio) of the two profiles to be mixed can be adjusted by the width of the "drag and drop" movement (defacto between 0 and 100%).By way of example, let us assume in this context that the first profile of the signal processing chain 30 was designed for natural hearing (home setting) and the third profile of the signal processing chain 30 was programmed as an autofocus for speech signals, so that the beamformer in the first signal processing stage 31 always selects a speech vector and the DSP (DSP stands for "Digital Signal Processor") of the third signal processing stage 33 is set to speech. If you now touch and hold the first profile zone 53 and then swipe downwards, the stable "home" profile approximates the uncompromising speech autofocus. Advantageously, however, the mixing ratio can be controlled by the width of the finger movement, i.e., dynamically adjusted in real time. As a result, an audio signal optimized for normal hearing is mixed with an audio signal optioned for speech according to the selected mixing ratio.

[0163] Using the "drag and drop" functionality, the user can also dynamically switch back and forth between the two profiles or dynamically change the mixing level as long as their finger is in the "drag" state on the touch-sensitive element but is being moved there. This can be done, for example, to determine which mixing ratio suits them best or to consciously switch back and forth between two mixing ratios in order to temporarily optimize their acoustic perception according to the first profile or the second profile at short intervals. The mixing ratio set at that time is only fixed when the user lifts their finger from the touch-sensitive element.

[0164] Furthermore, the user, i.e. the person 100 wearing the system 1, always returns to the original, stable listening mode (the "Home" profile) with a "swipe" gesture upwards or downwards, i.e. with a rapid swipe of the finger across the two zones touching the touch-sensitive element, regardless of whether this "swipe" gesture is performed on the first processing device 2 or the second processing device 3.

[0165] This makes it easy to control a complex command chain without an external device, such as a smartphone or a tablet computer, etc., which can intervene in each element of the signal processing chain 30 as necessary.

[0166] This allows the entire system 1 to be controlled via touch. As mentioned, the Touchbar 20 is freely assignable and can control virtually all system parameters, i.e., the settings of the signal processing chain 30. This allows the user to intervene in their sound reality in real time. As mentioned, dynamic profiles can also be generated, with the profiles placed on the Touchbar 20 being mixed with each other as discussed. This allows, for example, complex DSP functions to be easily controlled, and the user has the feeling of interacting with the sound. The feeling of the user simply changing programs with a plastic button, often perceived as a disadvantage with conventional systems, is thus completely avoided.

[0167] In addition, the user interface can adapt dynamically to the user, e.g. as a result of an interaction that has just been carried out or due to external circumstances that were determined, for example, through the evaluation of the incoming sound.

[0168] The user interface can also briefly change its state, for example, providing the function of a call answer button during the ringing of a smartphone call, which is communicated to System 1 via a Bluetooth connection, for example. However, the brief change can also be caused by the evaluation of the incoming sound, if this appears necessary based on the evaluation.

[0169] Likewise, a zone (e.g., the upper half) of the touch strip 20a or 20b can be assigned, for example, to the "Speech Autofocus" function. By tapping in the upper area of ​​the touch strip 20a or 20b assigned this menu item, a special plug-in (i.e., a software component) is then executed in processing stages 4 and 5, with the aid of which the sound field is searched for speakers and one or more virtual microphones are directed to these sound sources or spatial areas or directional areas in which the sound sources have been identified.

[0170] Similarly, a "simply listen" function can be activated via the Touchbar 20. This can be a completely stable listening program, but one that is adapted to the hearing curve of the System 1 user and is therefore suitable for or leads to a natural hearing experience. This function would consist of a mix of omnidirectional and / or, for example, cardioid directionally recorded audio signals (vector forward, front of user). The mix of the individual microphone signals MSI - MS4 is determined by binaural (localization) hearing tests. The frequency response (including compressor values) results from a binaural (frequency) hearing test. The polar pattern, i.e., the applied directional characteristic, is generated by mixing several virtual microphones. This applies to all programs of this System 1. The only difference here is that the mix is ​​absolutely static.The fact that the mix is ​​stable means that no automatic processes intervene during operation. Localization capability is restored through binaural mixdown, a process familiar from Ambisonics.

[0171] This "just listen" profile also forms the basis for all other profiles, with the difference that other profiles can create and freely mix freely definable virtual microphones of any type and number (limited to the order of the microphone array 10-13; in the present embodiment, it is First Order Ambisonics). One example of this is a "noise cancelling" function, which can be accessed on the touch strip 20a or 20b and has a dynamic behavior. When the "noise cancelling" profile is active, the environment is continuously analyzed, and background noise in the sound field is selectively removed fully automatically. This function searches for and suppresses permanent background noise, while simultaneously allowing music, ambient noise, and speech to be consistently perceived by the user. The signal processing chain 30 thus adapts dynamically to the spatial and temporal dynamics of the background noise.

[0172] In general, it can be stated that with an increasing number of microphones (and thus axes), the complexity / depth of the sound field sampling can be increased. This naturally allows for more precise calculations (narrower funnels of the virtual microphone, more accurate 3D field representation in the mixdown, more accurate energy distribution information / sound pressure vectors, etc.). The higher the Ambisonics order, the finer the content and spatial resolution of the system 1 becomes. This order increases with the number of microphones. Starting with a number of 4 microphones per device, i.e., per processing device 2 or 3 (2x4 / 1x8 matrix), each individual processing device 2 or 3 (i.e., both Herables) would enter the first order of the spherical harmonics.

[0173] The haptic module 45, which can also be referred to as a feedback module, provides feedback to the user, i.e., the person wearing or using the system 1, under a wide variety of operating conditions, for example, when the touch bar 20 or the respective touch strips 20a or 20b have been touched, or when the system 1 wants to communicate something during normal operation. This feedback module serves to enable uninterrupted "notifications" to the user. The sound image generated by the system 1 for the user is thus never disturbed by the haptic feedback, for example, when changing volume or program, because feedback is thus provided without acoustic interruption or acoustic overlay.

[0174] The function of system 1 is discussed below with reference to Figure 9 and the following figures, where Figure 9 shows an exemplary sound field arrangement in the environment of system 1. Figure 9 shows various sound sources 60, 61, 62, and 63, which have nothing to do with speech, around the user of system 1 (person 100), and five speakers 64, 65, 66, 67, and 68, respectively, who constitute sound sources for speech (here, the heads of speakers are symbolized, and their direction of view is indicated by the tips of the triangular symbols or arrows). Sound emitted by the various sound sources 60 to 68 is received by microphones 10 to 13 and can be automatically assigned in system 1 (speech or other sound events). Depending on the configuration, i.e. settings in the signal processing chain 30, the system 1 can focus on specific sound sources 60 to 68 using a virtual microphone.In a virtual microphone, a desired directional characteristic 69 is digitally defined via the signal processing chain 30. For example, in Figure 10A, the focus of the virtual microphone is placed on the two speakers 65 and 66 in the upper left quadrant (left front relative to person 100). In Figure 10b, in contrast, the focus is placed on the general sound source 60 located in the upper right quadrant (right front relative to person 100). The respective orientation of the "sound horn" of the virtual microphone thus determines the direction from which sound arriving at system 1 will be delivered to the ears of person 100, possibly under the influence of special signal processing measures in the signal processing chain 30.

[0175] The system 1 thus provides the function of combining the two processing devices 2 and 3 worn at the ear 102 and 103 of the person 100 by mutually transmitting the signals of all microphones distributed between the two processing devices 2 and 3 to form a single beamformer that uses all available microphones within the system 1.

[0176] The previously discussed example of a virtual microphone alignment can, in principle, be fully automated. One or more virtual microphones can be automatically aligned to speakers 65 to 68, and if necessary, they can also record speakers in groups, or, depending on the occurrence of speech from these directions, focus on the respective speakers 65 to 68 sequentially or simultaneously.

[0177] However, in order to allow the user, i.e. person 100, of system 1 full control over his or her hearing experience, a software application, or app for short, installed on a mobile device of the user (mobile user device, e.g. a smartphone 70) can be used, which communicates with system 1, e.g. via a Bluetooth connection.

[0178] The visual user interface of this app is illustrated by way of example in Figures 11A to 11D, with Figure 11A showing the user's smartphone 70 and the screen content 89. Here, three sound sources 72 to 74 are visualized within a circle 71, with the user, i.e. the person 100 using the system 1, being visualized in the center. Corresponding to the sound sources 72 to 74, sound direction sectors 72a, 73a, and 74a are visualized, which indicate the directional ranges from which the sound of the respective sound sources 72 to 74 is received. The sound direction sectors 72a, 73a, and 74a can be defined by a direction and a funnel width (angular sector). The funnel width can be changed or adjusted, for example, using a sliding element 78.can be adjusted, whereby individual sound sources 72 to 74 can be precisely isolated, i.e. ambient sound is largely blocked out, or ambient sound coming from the vicinity of the sound source 72 to 74 can be allowed in the audio signal to be further processed, or groups of neighboring sound sources 72 to 74 can be allowed together. Below this, three selection elements 79, 80 and 81 can be activated, with the help of which preset funnel widths can be directly selected. A 360° directional slider 75 can be moved along the circle 71 and is used to adjust the orientation in the sound field. A streaming element 76 can be activated to integrate an external audio streaming source (not shown) into the signal processing chain 30 of the system 1. This can be, for example, the smartphone 70 itself.A fixing element 77 can be activated to continuously fix at least one previously selected sound source 72 to 74, so that this selected sound source 72 to 74 remains in focus even when the user moves his head.

[0179] Figures 11B to 11D now show the selection of a sound source using the mobile phone 70, with the focus only on the essential screen content. In the present case, it is assumed that the user wishes to select the sound source 72, which is done by touching the area of ​​the screen displaying the sound source 72. As shown in Figure 11C, all other sound sources 73, 74 are now hidden (dashed). Figure 11D further visualizes the user modification options that the app makes available to the user for a selected sound source 72. Using the three displayed selection elements 82, 83, 84, the selected sound source 72 can be tracked angularly (selection element 82), suppressed (selection element 83), and isolated (selection element 84), whereby the button 84 is activated according to the user's intention, although this is not explicitly shown. Using the sliding element 78 orThe funnel width can now be set using one of the selection elements 79, 80 and 81, whereby the isolated sound source can be mixed with ambient sound according to the slider setting or a predefined funnel width (narrowly focused, semicircular, omnidirectional) is used.

[0180] The app thus allows the orientation as well as the parameterization of the virtual microphone and the parameterization of the further processing of the audio signal provided by the virtual microphone in the system 1. Thus, in a radio communication, the interaction data ID, which represents the settings defined with the help of the app, is communicated from the mobile phone to the processing devices 2 and 3, where the signal processing chain 30 is set accordingly.

[0181] The app discussed here can be understood as an "environment tool." The app utilizes the spherical sound recording, analysis, and output capabilities of System 1 to analyze, process, and output the sound field and its objects, allowing manual user interaction.

[0182] Furthermore, it should be mentioned that the position data LD generated by the gyroscope 21a and / or 21b can also be transmitted to the smartphone 70 and thus to the app running there in order to correctly convert the head movements into rotations of the screen content, i.e. to reproduce the real conditions regarding the orientation.

[0183] The functionality described above is explained below using an application example. Let's assume that a hearing-impaired user wearing System 1 is sitting in a busy restaurant. It's loud, many people are talking, and the noise of the air conditioning and the hectic kitchen fills the (sound I) room. It's very difficult (without System 1) to understand the person sitting next to them. The user needs their full concentration to participate in the conversation and can't truly enjoy their experience at the restaurant. The user then opens the "Environment Tool," i.e., the app. The app receives the sound field information acquired by System 1 from System 1 via a Bluetooth connection, analyzes the scene, and graphically visualizes the detected scene, as shown in Figure 11A. The user is familiar with the app's user interface and uses it to optimize their sound perception.For example, they tap on the screen their neighbor (sound source 72). The user thereby defines the instruction to point a virtual microphone at the desired sound source 72. Since the desired sound source 72 contains a speech signal, the signal processing (of the signal processing chain 30) is automatically optimized for speech understanding.

[0184] Alternatively, as mentioned above, system 1 can be operated in an automatic mode. In this automatic mode, the user can consent to an automatically calculated modification of the sound perception through interaction (such as tapping one of the processing devices 2 or 3). Alternatively, system 1 can also be operated fully autonomously, whereby no user interaction with system 1 would be necessary.

[0185] Ultimately, this means that the user can now understand their neighbor better, is no longer disturbed by ambient noise, and can finally enjoy their restaurant experience because it is now much easier for them to understand their neighbor and have a fluent conversation with them.

[0186] The sound output artificially generated using system 1 can also exhibit a higher signal-to-noise ratio than the human sense of hearing. This depends in particular on the number of microphones. Therefore, it is advantageous to use more than two microphones per processing device 2 or 3. A spherical microphone array, with sufficient resolution, which is associated with the number of microphones, can overshadow the normal human sense of hearing in terms of its ability to isolate individual sound sources.

[0187] The extraction of virtual microphone signals is of enormous value, especially for people with hearing impairments, as this method represents a new approach to resolving "I don't understand you" situations. For the first time, it is possible to process multiple sound sources in a sufficiently short time with the goal of understanding a conversation partner properly in a room with a very challenging sound environment, such as a cocktail party (complex / noisy / multi-layered sound environments), where it is difficult to understand anything. In this environment, one virtual microphone can easily focus on the desired conversation partner, or several virtual microphones can even focus on a large number of conversation partners, solving the problem of speech understanding in such complex sound environments.

[0188] In the following, an internet connection of System 1 and its various aspects are discussed with the help of Figure 12. It should be clarified that, for reasons of clarity, not all reference symbols have been included in this illustration.

[0189] System 1 can be connected to the Internet 86, which is visualized using a cloud, via a single system-external radio connection 85A. If only one of the two processing devices 2 or 3 is connected to the Internet 86, it is necessary for the two processing devices 2 and 3 to be synchronized with each other if necessary, which can be achieved, for example, by a system-internal radio connection 85B between the two processing devices 2 or 3. Preferably, however, both processing devices 2 or 3 can be connected to the Internet 86, for which a second system-external radio connection 85C is used. The system-external radio connections 85A and 85C as well as the system-internal radio connection 85B can be implemented using the radio modules 6 to 9 or their interaction.

[0190] However, the connection to the Internet 86 can also be made via a cable via the appropriately designed extension module 51.

[0191] Via the internet 86, the system 1 can exchange data with a server 87 accessible via the internet 86. This functionality can be used for a wide variety of applications, which are listed below in a non-exhaustive manner. Furthermore, it should be noted that the aforementioned system-external radio connections 85A and 85C can also be implemented using the user's aforementioned mobile device 70 (smartphone or tablet computer), in which case these system-external radio connections 85A and 85C are replaced by the internet connectivity 88 of the mobile device 70.

[0192] For example, the system 1 can store user (identification) data relating to the user of the system 1 and / or system (identification) data relating to the system 1 on the server 87 in order to, for example, link the processing devices 2 or 3 in a unique manner to the user in question.

[0193] However, the server 87 can also be used purely as a data storage device, with the processing devices 2 and 3, for example, temporarily storing raw data of the recorded sound event on the server 87 in order to also carry out the signal processing running in the processing devices 2 and 3 on the server side and, if necessary, to further develop or optimize it. The same applies to the audio signal to be delivered to the user of the system 1, the underlying fourth audio data stream AD4 of which can also be delivered to the server 87 for the aforementioned purposes. The intermediate results of the signal processing chain 30, namely the preceding three audio data streams AD1 to AD3, can also be processed analogously.

[0194] However, server 87 can also be used to provide a so-called plug-in store. This is understood to be a software distribution system for the controlled distribution of software and / or audio data modules, whereby these software or audio data modules are delivered specifically to certain processing devices 2 or 3, possibly on a user-specific basis, possibly only against payment, possibly only against a doctor's prescription, etc.

[0195] The server 87 can also be used to form a gateway to other digital services or service providers. For example, third-party providers can digitally access the processing devices 2 or 3 to conduct tests. These tests can be, for example, hearing tests conducted by audiologists, health insurance companies, or (health) insurers, either manually or fully automatically. However, these tests can also relate to the technical functionality of system 1, i.e., the two processing devices 2 or 3, which are used in the event of a suspected malfunction, a malfunction, or even a routine check of the processing devices 2 or 3, preferably fully automatically.

[0196] The server 87 can also serve as a source for a health database into which the hearing characteristics of the affected person 100 determined by the processing devices 2 or 3 are imported in order to automatically complete the overall health profile of the person 100 and, in particular, to document it over time. Conversely, for example, when replacing one of the processing devices 2 or 3, the previously determined hearing characteristics of the person 100 can be loaded from the server 87, and a basic setting of the signal processing chain 30 of the new processing device 2 or 3 can be made.

[0197] Server 87 can also be used as an "audio logistics interface" (communication, secure programming line). This involves providing certified entities (doctors / audiologists) with a special communication channel to processing devices 2 and 3, respectively, to influence parameters reserved for specialized personnel. These could, for example, be parameters that are potentially hazardous to hearing and should only be operated by trained specialists.

[0198] The connection to the Internet 86 and the server services allow a variable and individual extension of the functionalities of the system 1.

[0199] Although the function of System 1 has so far been discussed using a two-dimensional polar chart based on the directional characteristics plotted there for the sake of simplicity, it should be noted here that the directional characteristics have a three-dimensional character. Thus, any virtual microphone can also be aligned or oriented three-dimensionally (within certain limits, if necessary).

[0200] It should also be noted that System 1 can also be operated in emergency mode if the radio connection(s) between the two processing devices 2 and 3 are not present or are interrupted. Each of the processing devices then continues to operate autonomously, but the system no longer allows binaural hearing.

[0201] Furthermore, it should be mentioned that the position data LD generated by the gyroscope 21a and / or 21b can also be transmitted to a general computer (desktop, laptop, tablet computer, etc.), where it can be used to control a screen pointer. The system 1 coupled to the computer thus simulates a pointing device, such as a laser pointer, which is guided by the hand in order to convert head movements into movements of a virtual laser point on the computer screen. Finally, it is pointed out once again that the figures described in detail above are only exemplary embodiments that can be modified in a variety of ways by a person skilled in the art without departing from the scope of the invention. For the sake of completeness, it is also pointed out that the use of the indefinite articles "a" or "an" does not exclude the possibility that the features in question may be present multiple times.

Claims

Claims 1. System (1) for processing microphone signals (MSI, MS2, MS3, MS4), comprising: - a first processing device (2) for providing a first group (Gl) of individual microphone signals (MS1, MS2) and - a second processing device (3) for providing a second group (G2) of individual microphone signals (MS3, MS4), wherein the first processing device (2) is designed to transmit the first group (G1) of microphone signals (MS1, MS2) via a first radio connection (15) from the first processing device (2) to the second processing device (3), which is designed to receive the first group (G1) of microphone signals (MS1, MS2), and wherein the second processing device (3) is designed to transmit the second group (G2) of microphone signals (MS3, MS4) via a second radio connection (16) separate from the first radio connection (15) from the second processing device (3) to the first processing device (2), which is designed to receive the second group (G2) of microphone signals (MS3, MS4).

2. System (1) according to claim 1, which forms a binaural hearing aid system (1), wherein - the first processing device (2) for wearing at the position of the left ear (102) of a head (101) of a person (100) and wherein - the second processing device (3) is designed to be worn at the position of the right ear (103) of the head (101) of the person (100).

3. System (1) according to one of the preceding claims, wherein at least one processing device (2), preferably both processing devices (2, 3), each has an output module (22), in particular a loudspeaker (23a, 23b) and / or an output module for controlling an implant.

4. System (1) according to one of the preceding claims, wherein - the first processing device (2) has a first radio module (6) for transmitting the first group (Gl) of microphone signals (MSI, MS2) and the second processing device (3) has a second radio module (7) for receiving the first group (Gl) of microphone signals (MSI, MS2) and wherein - the second processing device (3) has a third radio module (8) for transmitting the second group (G2) of microphone signals (MS3, MS4) and the first processing device (2) has a fourth radio module (9) for receiving the second group (G2) of microphone signals (MS3, MS4).

5. System (1) according to one of the preceding claims, wherein the two processing devices (2, 3) are designed to provide a permanent first and second radio connection (15, 16).

6. System (1) according to one of the preceding claims, wherein - the first processing device (2) is designed to process the first group (G1) of microphone signals (MS1, MS2) together with the second group (G2) of microphone signals (MS3, MS4) received via radio connection in a phase-synchronized manner, and wherein - the second processing device (3) is designed to process the second group (G2) of microphone signals (MS3, MS4) together with the first group (G1) of microphone signals (MS1, MS2) received via radio connection in a phase-synchronous manner.

7. System (1) according to one of the preceding claims 4 to 6, wherein the first radio module (6) is designed to receive at least a first microphone signal (MS1) from a first microphone (10) and a second microphone signal (MS2) from a second microphone (11) and to transmit them as the first group (G1) of individual microphone signals (MSI, MS2) and the third radio module (8) is designed to receive at least a third microphone signal (MS3) from a third microphone (12) and a fourth microphone signal (MS4) from a fourth microphone (13) and to transmit them as to transmit the second group (G2) of individual microphone signals (MS3, MS4).

8. System (1) according to one of the preceding claims, wherein each of the processing devices (2, 3) has a processing module which is designed to process the two groups (G1, G2) of the microphone signals (MS1, MS2; MS3, MS4).

9. System (1) according to one of the preceding claims, wherein each of the processing devices (2, 3), in particular each of the processing modules according to claim 8, is designed to process the two groups (G1, G2) of microphone signals (MSI, MS2; MS3, MS4) according to the Ambisonics format. preceding claims, wherein at least one of the processing devices (2), preferably both processing devices (2, 3), has a position sensor (21a, 21b) for providing a position sensor signal, preferably a mechanical position sensor, particularly preferably an acceleration sensor and / or a gyroscope (21a, 21b), and is designed to process the microphone signals (MSI, MS2, MS3, MS4) taking the position sensor signal into account.

11. System (1) according to one of the preceding claims, wherein at least one processing device (2), preferably both processing devices (2, 3), has an input module (20) for detecting a user interaction and the processing device (2, 3) is designed to process the microphone signals (MS1, MS2, MS3, MS4) taking into account the detected user interaction.

12. System (1) according to claim 11, wherein the input module (20) is designed to detect user interactions within the auricle or in front of the auricle or along the front edge of the auricle, in particular towards the upper edge.

13. System (1) according to one of claims 11 to 12, wherein the input module (20) has at least one touch bar or is designed as a touch bar.

14. System (1) according to one of the preceding claims 11 - 13, wherein a housing (38), on which a touch-sensitive surface of a first touch bar is accessible, is connected at least on one side, but preferably on both sides, to a deformable, in particular plastically deformable, connecting element (37, 40).

15. System (1) according to claim 14 in combination with claim 3, wherein the deformable connecting element as a first connecting element (37) connects a housing of a signal-emitting part of the output module (22) to a housing (38) on which the touch-sensitive surface of the first touch bar is accessible.

16. System (1) according to claim 8, wherein the processing module comprises a housing (41), in particular a rigid housing, which housing (41) is designed to be worn behind the auricle, in particular which housing (41) is adapted to the course of the auricle from the upper region to the rear region of the auricle.

17. System (1) according to one of claims 15 - 16, wherein the housing (41) of the processing module is connected at least on one side to a deformable, in particular plastically deformable, connecting element (40).

18. System (1) according to claim 17, wherein the deformable connecting element as a second connecting element (40) connects the housing (41) of the processing module to the housing (38) on which the touch-sensitive surface of the first touch bar is accessible.

19. System (1) according to one of claims 16 - 18, wherein at least one of the processing devices (2, 3) has a wired connection (43), preferably a multi-pole connection, in particular a magnetically holding connection for connecting another device or an additional module (51).

20. System (1) according to one of the preceding claims, wherein at least one of the processing devices (2, 3) has a haptic module (45) (also referred to as haptic module (45)) for emitting a haptically perceptible signal.

21. System (1) according to one of the preceding claims, wherein at least one of the processing devices (2, 3) is designed to receive and process an additional software application or audio output data representing an audio signal that can be specified with the aid of the processing device (2, 3).

22. System (1) according to claim 21, wherein the two processing devices (2, 3) are designed to transmit the software application received only at one of the processing devices (2, 3) or the audio output data received only at one of the processing devices (2, 3) to the other processing device (3, 2) by means of a third radio connection.

23. System (1) according to one of the preceding claims, wherein at least one of the processing devices (2, 3) is designed to communicate with a mobile user device (70) via a fourth radio connection - and to provide position information to said user device (70) by means of the fourth radio connection, wherein the position information indicates the position of at least one sound source (60-68, 72-74) in relation to the position of the system (1), - and to receive selection information from this user device (70) using the fourth radio connection, the selection information indicating which direction or which directional range is selected with respect to the position and / or orientation of the system (1) for the further processing of the microphone signals (MS1, MS2, MS3, MS4).

24. A first method for operating a system (1) for processing microphone signals (MS1, MS2, MS3, MS4), the first method comprising the following method steps, namely: - providing a first group (Gl) of individual microphone signals (MS1, MS2) with a first processing device (2) and - providing a second group (G2) of individual microphone signals (MS3, MS4) with a second processing device (3), wherein with the aid of the first processing device (2) the first group (G1) of microphone signals (MS3, MS4) is transmitted via a first radio connection (15) from the first processing device (2) to the second processing device (3), which is designed to receive the first group (G1) of microphone signals (MS3, MS4), and wherein with the aid of the second processing device (3) the second group (G2) of microphone signals (MS3, MS4) is transmitted via a second radio connection (16) separate from the first radio connection (15) from the second processing device (3) to the first processing device (2), which is designed to receive the second group (G2) of microphone signals (MS3, MS4).

25. Second method for operating a processing device (2), the second method comprising the following method steps, namely: - providing a group (Gl) of individual microphone signals (MS1, MS2) present at the processing device (2) via a first radio connection (15) and - receiving a group (G2) of individual microphone signals (MS3, MS4) present at another processing device (3) via a second radio connection (16) separate from the first radio connection (15).

26. A computer program product comprising software code adapted to carry out the method steps according to the second method according to claim 25 on a programmable hardware of the processing device (2).

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